tag:blogger.com,1999:blog-52060969334204936442024-03-20T08:09:53.926-07:00Telephony- SIP, VoIP, IMS, SIPPTechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.comBlogger12125tag:blogger.com,1999:blog-5206096933420493644.post-81679018553706487472015-04-29T01:07:00.002-07:002015-04-29T01:07:29.850-07:00Pure SIP<div dir="ltr" style="text-align: left;" trbidi="on">
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<span style="color: white;"><span style="font-family: Verdana, sans-serif;"><b><span style="background: #808000;">---
SIP Overview --- </span></b></span></span></div>
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<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q0"></a><span style="color: black;"><b>Where
do we experience Internet Telephony or/and SIP in our daily lives?</b>
</span>
</span><br />
<span style="color: black;"><span style="font-family: Verdana, sans-serif;">We
use these technologies in many different ways. Sometimes we are not
even aware of using it... For instance one may use a calling card
number to make a phone call, which routes the call via VoIP gateways.
Instant Messaging tools use these technologies. Some people use IP
phones (e.g. Vonage) at home. Some use PTT phones which utilize VoIP
technology. There are many more areas/ways where we use these
technologies. The key areas are illustrated in our eLearning.</span></span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q1"></a><span style="color: black;"><b>SIP
is a VoIP protocol – so does it carry voice packets?</b> </span>
</span><br />
<span style="color: black;"><span style="font-family: Verdana, sans-serif;">For
the most part SIP does signaling. Voice is carried by other real time
protocols such as RTP. There are some implementations however where
SIP is used to carry the media itself. For example SIP can carry the
media (normally text) of an Instant Message in the payload of a SIP
message (request) called MESSAGE...</span></span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2"></a><span style="color: black;"><b>Do
I always need to use a proxy server?</b> </span>
</span><br />
<span style="font-family: Verdana, sans-serif;">First of
all it is not you, it is rather your SIP phone that may need to use
it... Second, the answer is NO. SIP phone needs to use SIP proxy
server only when it does not know the IP address (or host name) of
the destination or if the policy of the operator (ISP) mandates it.
So for example say you like to establish a white board session with a
colleague of yours, using SIP based client such as Net Meeting. Now
if she provides you the IP address of her machine, you can contact
her machine directly.
</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3"></a><span style="color: black;"><b>Is
the Microsoft messenger the only SIP based messenger tool?</b> </span>
</span><br />
<span style="color: black;"><span style="font-family: Verdana, sans-serif;">Definitely
not.</span></span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q4"></a><span style="color: black;"><b>I
want to build a SIP phone - is SIP all what it takes?</b> </span>
</span><br />
<span style="color: black;"><span style="font-family: Verdana, sans-serif;">Well,
not quite. If you don't equip it with RTP stack and couple of
vocoders it might be useless. It depends of course on the use you
intend for it.</span></span><br />
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<span style="color: white;"><span style="font-family: Verdana, sans-serif;"><b><span style="background: #808000;">---
SIP Functionality ---</span></b></span></span></div>
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<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q1.1"></a><b>Does
SIP support the standard telephone features?</b></span></div>
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<br /></div>
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<span style="font-family: Verdana, sans-serif;">Yes.
SIP supports, among others:</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;"><br />*
call forwarding unconditional, busy, ...<br />* call transfer (call
control spec)<br />* caller ID<br />* call hold<br />* 3-way conferences
and multiparty conferencing (call control spec)<br />* call return
("*69")<br />* call park (with NOTIFY)<br />* follow-me<br />*
find-me<br />* call waiting<br />* IVR systems<br />* multiple line
presences<br />* call waiting<br />* camp on<br />* call queueing<br />*
automatic call distribution<br />* do not disturb </span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Some
services, like repetitive dialing, station speed dialing, last number
redial, and distinctive ringing, are implemented purely in the end
system and require no support from the signaling protocol. The
Telecommunications Industry Association (TIA) is working on a
<a href="http://web.archive.org/web/20030213031908/http://www.tiaonline.org/standards/ip/">recommendation</a>
for business PBX-style services and other Internet phone
requirements.</span></div>
<div style="line-height: 100%;">
<br /></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q1.2"></a><b>How
does SIP support caller ID?</b></span></div>
<div style="line-height: 100%;">
<br /></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Caller-ID
is provided by the <i>From</i> SIP header containing the caller's
name and "number". The number would most likely be placed
in the user field of a SIP URL or appear in a tel: URL. Since the
callee generally does not know or trust the callee's server, only
cryptographic signatures can be used to ensure that the information
is valid. For example, the outgoing proxy might be operated by an
ISP, enterprise or phone company and sign for the identity of the
caller, using the <i>signedby</i> parameter, with the identity of the
company verified by a public key certificate similar to those used by
web sites. </span></div>
<div style="line-height: 100%;">
<br /></div>
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<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q1.3"></a><b>Should
SIP be used to join a conference from a web page?</b></span></div>
<div style="line-height: 100%;">
<br /></div>
<span style="font-family: Verdana, sans-serif;">It is possible to embed a
SIP URL in a web page, including a session description. Clicking on
that link triggers an invitation for the conference listed to the
address contained in the URL. Unfortunately, the current standard
browsers (Netscape and Internet Explorer) make it difficult or
impossible to add support for another URL type. Until SIP URIs are
implemented in standard browsers, <a href="http://web.archive.org/web/20030213031908/http://www.cs.columbia.edu/~hgs/html/data.html"><i>data:</i></a>
URLs can be used to implement similar functionality, albeit less
elegantly. If it is desired that following the link directly adds the
user to an existing conference, e.g., for a conference "TV
guide"-style directory, the <a href="http://web.archive.org/web/20030213031908/http://www.normos.org/rfc/rfc2397.txt"><i>data:</i></a><a href="http://web.archive.org/web/20030213031908/http://www.normos.org/rfc/rfc2397.txt">
URL</a> is more appropriate.
</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q1.4"></a><b>Can a
SIP-initiated session have zero or one participants?</b></span><br />
<span style="font-family: Verdana, sans-serif;">SIP-initiated sessions can
have no or just one participant. Examples of a session with no
participants include an invitation to a multicast group with no
members (beyond the invited party). Also, SDP sessions can start at a
future time relative to the invitation.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q1.5"></a><b>How
do I charge/bill for Internet telephony using SIP?</b></span><br />
<span style="font-family: Verdana, sans-serif;">This depends on whether you
plan to charge for SIP services like directory look-ups, call
processing or mobility, for gateway services to the PSTN, or for
carrying media data:
</span><br />
<span style="font-family: Verdana, sans-serif;">SIP services - </span><br />
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">The
<var>Authorization</var>
header can be used to indicate a customer identity that associates a
SIP request with a billable entity.
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Examples
of possibly chargeable SIP services include:</span></div>
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<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image1" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
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<span style="font-family: Verdana, sans-serif;">Directory
services such as SIP proxy/redirect lookups;
</span><br />
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<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image2" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
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<span style="font-family: Verdana, sans-serif;">Customer
profile management;</span><br />
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<span style="font-family: Verdana, sans-serif;">SIP
server operations can be charged based on server logs or, for
real-time billing, via AAA.</span></div>
<span style="font-family: Verdana, sans-serif;">Media services - </span><br />
<span style="font-family: Verdana, sans-serif;">Media services include
retrieving and storing voice mail, as well as transcoding of media
streams. They are not initiated by SIP, but, for example, via <a href="http://web.archive.org/web/20030213031908/http://www.cs.columbia.edu/~hgs/rtsp">RTSP</a>.</span><br />
<span style="font-family: Verdana, sans-serif;">Gateway services - </span><br />
<span style="font-family: Verdana, sans-serif;">Similar to SIP services.
Care has to be taken to stop billing when (say) RTP voice data is no
longer flowing through the gateway. The gateway will generate call
detail records (CDRs) either directly or through RADIUS.
</span><br />
<span style="font-family: Verdana, sans-serif;">Transport (network
services) - </span><br />
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">It
seems unlikely that voice calls carried over a best-effort service
will generate per-minute charges. When reserving bandwidth or
guaranteeing other quality-of-service parameters, the resource
reservation protocol or differentiated services are the appropriate
mechanism for including charging. These reservation protocols will
likely be used in applications that are not initiated by SIP, for
example, audio/video on demand or VPNs. Actual accounting records may
be generated by AAA protocols (e.g., by policy enforcement points
(PEP) or policy decision points (PDP)) or log files.
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Under
some circumstances, a SIP proxy server may be useful to initiate such
reservations or differentiated services treatment on behalf of a
call, since it may be easier to authenticate the SIP request than the
lower-layer reservation request or the end system may not be capable
of making reservations or marking packets. In those cases, the SIP
proxy would initiate a resource reservation and "charge back"
the caller identified by the SIP request.</span></div>
<div style="line-height: 100%;">
<br /></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q1.6"></a><b>How
do prepaid calling cards work in SIP?</b></span></div>
<div style="line-height: 100%;">
<br /></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Note
that, in general, prepaid calling cards only make sense in an IP
network if there is a special-purpose VoIP internet, calls traverse a
IP-to-PSTN gateway or VoIP packets receive special treatment. The SIP
requests are forced to traverse a stateful proxy, which controls the
Internet telephony gateway, router QOS function or firewall,
depending on the architecture. When the time is used up, the proxy or
gateway issues a <var>BYE</var>
request to both parties, using the existing call ID. It also disables
the gateway connection, turns of any special QOS treatment for the
RTP packets or closes the firewall for that stream. This requires no
additions to either caller or callee. Relying on SIP <var>BYE</var>
itself only suffices the end systems can be trusted by the network
provider not to keep sending packets.</span></div>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q1.7"></a><b>Does
SIP carry DTMF? </b></span><br />
<span style="font-family: Verdana, sans-serif;">There are at least two
options for carrying DTMF and similar signals in a VoIP network using
SIP. First, DTMF can be transported as an RTP payload (<a href="http://web.archive.org/web/20030213031908/http://www.normos.org/rfc/rfc2833.txt">RFC
2833</a>). This has the advantage that it provides accurate timing
and alignment with the speech RTP packets. Also, media gateways are
the most likely to detect and generate tones, so that making it part
of the media stream is appropriate. However, under some
circumstances, it may be necessary for signaling entities to know
about DTMF signals. Currently, there is no standardized solution
within SIP, but it has been proposed to carry DTMF information in SIP
INFO messages, either encoded as simple text or using the RFC 2833
format. The latter is more complex, but offers duration and timing
information.</span><br />
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<span style="font-family: Verdana, sans-serif;"><span style="color: white;"><b><span style="background: #808000;">---
SIP Protocol Operation</span></b></span><span style="color: white;"><b><span style="background: #808000;">
</span></b></span><span style="color: white;"><b><span style="background: #808000;">---</span></b></span></span></div>
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<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.1"></a><b>What
does the [H14.17] in RFC 3261 stand for?</b></span><br />
<span style="font-family: Verdana, sans-serif;">This is explained in
Section 3 of RFC 2543. It refers to the section number in the
HTTP/1.1 specification.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.2"></a><b>Do
callers need to know the location of the location server?</b></span><br />
<div style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">The
caller doesn't interact with the location server directly. A redirect
or proxy server asks the location server (which may be co-resident
with the SIP server or not) for "advice". The location
server is just a logical abstraction to indicate where the SIP server
gets its information from. The protocol between SIP server and
location server is beyond the scope of SIP. Examples of location
servers include
</span></div>
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<span style="font-family: Verdana, sans-serif;">finger;
</span><br />
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<span style="font-family: Verdana, sans-serif;">LDAP/X.500;
</span><br />
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<span style="font-family: Verdana, sans-serif;">whois,
whois++;
</span><br />
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<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image6" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
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<span style="font-family: Verdana, sans-serif;">ph
and other local directories;
</span><br />
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<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image7" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
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<span style="font-family: Verdana, sans-serif;">shared
file systems with registration on login;
</span><br />
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<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image8" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
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<span style="font-family: Verdana, sans-serif;">local
SQL databases reached through TCP.</span><br />
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<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Also,
callers don't register with the location server.
</span></div>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.3"></a><b>Which
parts of SIP are case-sensitive or case-insensitive?</b></span><br />
<table cellpadding="2" cellspacing="0" style="width: 247px;">
<colgroup><col width="209"></col>
<col width="27"></col>
</colgroup><tbody>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: 1px double #000000; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0.05cm;" width="209">
<span style="font-family: Verdana, sans-serif;">Method
</span><br />
</td>
<td style="border: 1px double #000000; padding: 0.05cm;" width="27">
<span style="font-family: Verdana, sans-serif;">CS
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="209">
<span style="font-family: Verdana, sans-serif;">Header
field name
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="27">
<span style="font-family: Verdana, sans-serif;">CI
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="209">
<span style="font-family: Verdana, sans-serif;"><var>Hide</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="27">
<span style="font-family: Verdana, sans-serif;">CI
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="209">
<span style="font-family: Verdana, sans-serif;"><var>Accept-Encoding</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="27">
<span style="font-family: Verdana, sans-serif;">CI
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="209">
<span style="font-family: Verdana, sans-serif;"><var>Accept</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="27">
<span style="font-family: Verdana, sans-serif;">CI
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="209">
<span style="font-family: Verdana, sans-serif;"><var>Accept-Language</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="27">
<span style="font-family: Verdana, sans-serif;">CI
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="209">
<span style="font-family: Verdana, sans-serif;">Encoding
name (PCMU, L16, etc.)
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="27">
<span style="font-family: Verdana, sans-serif;">CI
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="209">
<span style="font-family: Verdana, sans-serif;">rfc1123-date
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="27">
<span style="font-family: Verdana, sans-serif;">CS
</span><br />
</td>
</tr>
</tbody></table>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.4"></a><b>What
is the difference between a call leg and a call id?</b></span><br />
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">A
call leg refers to the one-to-one signaling relationship between two
user agents (UAs). The <var>Call-ID</var>
is an identifier, carried in the SIP messages, that refers to the
call. A call is a collection of call legs. A UAC starts by sending an
<var>INVITE</var>;
because of forking, it may receive multiple 200 OKs from different
UAs. Each corresponds to a different call leg within the same call.
Call is thus a grouping of call legs. In the call control spec,
additional call legs are created through the <var>Also</var>
header.
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Call
legs refer to end-to-end connections between user agents, rather than
any relationship with proxies. Within a call leg, there are numerous
transactions in both directions.
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">The
request URI is not used in call leg identification.</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">The
<var>To</var>
and <var>From</var>
field relate to local and remote in the following way. When Alice
sends a request on a call leg to Bob, the <var>From</var>
field contains the local address (Alice), and the <var>To</var>
field the remote address (Bob). When a request is received by Bob,
the <var>To</var>
field is matched to Bob's local address, and the <var>From</var>
field to the remote address (Alice).</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">The
<var>CSeq</var>
spaces in the two directions of a call leg are independent. Within a
single direction, the sequence number is incremented for each
transaction.</span></div>
<div style="line-height: 100%;">
<br /></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.5"></a><b>What
is the difference between tag and branch-id?</b></span></div>
<div style="line-height: 100%;">
<br /></div>
<span style="font-family: Verdana, sans-serif;">Branch IDs allow proxies to
match responses to forked requests. Without them, a proxy wouldn't be
able to tell which branch a response corresponds to. Tags, in <var>To</var>
headers, are of no help here since they are not known until responses
arrive. Tags are used by the UAC to distinguish multiple final
responses from different UAS.
</span><br />
<span style="font-family: Verdana, sans-serif;">A UAS has no reliable way
of determining if the request has been forked or not. Thus, to be
safe it needs to add a tag. Proxies only insert tags into the final
responses they generate themselves; they never insert tags into
requests or responses they forward.
</span><br />
<span style="font-family: Verdana, sans-serif;">Since a request can be
forked several times on its way to UAS, a single "tag" (or
whatever you like to call it) added to the request by one of the
proxies is not sufficient for the next forking proxy along the chain
to match responses on its own branches; every proxy that forked the
request would need to add its own unique IDs to the branches it
created. This is precisely what's being achieved by the branch
parameter in the <var>Via</var>
header. (Igor Slepchin)
</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.6"></a><b>How
can one recognize a retransmitted request?</b></span><br />
<table cellpadding="2" cellspacing="0" style="width: 643px;">
<colgroup><col width="75"></col>
<col width="92"></col>
<col width="65"></col>
<col width="393"></col>
</colgroup><tbody>
<tr>
<th style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: 1px double #000000; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0.05cm;" width="75">
<div align="left">
<span style="font-family: Verdana, sans-serif;">header
</span></div>
</th>
<th style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: 1px double #000000; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0.05cm;" width="92">
<span style="font-family: Verdana, sans-serif;">retransmitted
</span><br />
</th>
<th style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: 1px double #000000; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0.05cm;" width="65">
<span style="font-family: Verdana, sans-serif;">duplicate
</span><br />
</th>
<th style="border: 1px double #000000; padding: 0.05cm;" width="393">
<div align="left">
<span style="font-family: Verdana, sans-serif;">matching
response
</span></div>
</th>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="75">
<span style="font-family: Verdana, sans-serif;"><var>From</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="92">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="65">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="393">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="75">
<span style="font-family: Verdana, sans-serif;"><var>To</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="92">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="65">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="393">
<span style="font-family: Verdana, sans-serif;">same,
but tag may have been added
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="75">
<span style="font-family: Verdana, sans-serif;"><var>Call-ID</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="92">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="65">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="393">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="75">
<span style="font-family: Verdana, sans-serif;"><var>request
URI</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="92">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="65">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="393">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="75">
<span style="font-family: Verdana, sans-serif;"><var>CSeq</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="92">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="65">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="393">
<span style="font-family: Verdana, sans-serif;">same
</span><br />
</td>
</tr>
<tr>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="75">
<span style="font-family: Verdana, sans-serif;"><var>Via</var>
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="92">
<span style="font-family: Verdana, sans-serif;">-
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: none; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0cm; padding-top: 0cm;" width="65">
<span style="font-family: Verdana, sans-serif;">-
</span><br />
</td>
<td style="border-bottom: 1px double #000000; border-left: 1px double #000000; border-right: 1px double #000000; border-top: none; padding-bottom: 0.05cm; padding-left: 0.05cm; padding-right: 0.05cm; padding-top: 0cm;" width="393">
<span style="font-family: Verdana, sans-serif;">must
be local host; check for branch parameter to identify which branch
</span><br />
</td>
</tr>
</tbody></table>
<div style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">Looped
request are recognized by one or more of the following:
</span></div>
<table cellpadding="0" cellspacing="0" style="width: 100%px;">
<colgroup><col width="16*"></col>
<col width="240*"></col>
</colgroup><tbody>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image9" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">The
server finds itself in the request's <var>Via</var>
list, <em>including
any branch parameter</em>.
(The server should compute the branch parameter so that it depends
on the request URI.)
</span><br />
</td>
</tr>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image10" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">The
server is about to proxy the request to one of the hosts listed in
the <var>Via</var>
list. The same
</span><br />
</td>
</tr>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image11" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">The
<var>Max-Forward</var>
count is decremented to zero.
</span><br />
</td>
</tr>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image12" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">The
<var>Expires</var>
time has elapsed.</span><br />
</td>
</tr>
</tbody></table>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.7"></a><b>How
does a caller find its local registrar?</b></span><br />
<span style="font-family: Verdana, sans-serif;">The local registrar is
either manually configured or discovered via DHCP (<a href="http://www.rfc-editor.org/rfc/rfc3361.txt">RFC
3361</a>) . Another more theoretical option is: the SIP client issues
a multicast registration request to the sip.mcast.net
standard
multicast address, which all registrars (are supposed to) listen to
(but in practice not all do).</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.8"></a><b>Is
the domain of the request-URI and the To header always the same?</b></span><br />
<span style="font-family: Verdana, sans-serif;">The Request-URI names the
destination of the registration request, i.e., the domain of the
registrar. The user name must be empty. Generally, the domains in the
Request-URI and the To header field have the same value; however, it
is possible to register as a "visitor", while maintaining
one's name. For example, a traveler sip:alice@acme.com (<var>To</var>)
might register under the Request-URI sip:atlanta.hiayh.org with the
former as the <var>To</var>
header field and the latter as the Request-URI. Note, however, that
requests for a user at acme.com are not likely to arrive at the
atlanta.hiayh.org server; special purpose routing logic will
generally need to be established in order for requests for
alice@acme.com to go to the atlanta.hiayh.org server. In the vast
majority of cases, the domains in the request URI and To field will
match. The <var>REGISTER</var>
request is no longer forwarded once it has reached the server whose
authoritative domain is the one listed in the Request-URI.</span><br />
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.9"></a><b>Are
ACK requests retransmitted?</b></span></div>
<div style="line-height: 100%;">
<br /></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Not
per say. An <var>ACK</var>
is sent when a response retransmission is received. Reliability is
achieved because the response is retransmitted until an <var>ACK</var>
arrives, and the <var>ACK</var>
is retransmitted on response retransmissions. <var>ACK</var>
is only used for <var>INVITE</var>.</span></div>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.10"></a><b>How
are BYE requests routed?</b></span><br />
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Since
a <var>Contact</var>
header MUST be present in <var>INVITE</var>
and 200, the <var>BYE</var>
will go directly to the user agent if there is no <var>Record-Route</var>
header. If there is a <var>Record-Route</var>,
it will traverse the list of proxies indicated there.
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">If
the caller decides to send a <var>BYE</var>
before receiving a 200 from the callee, the <var>BYE</var>
is being handled by the proxies just as the corresponding <var>INVITE</var>
was handled, i.e., it may be forked.
</span></div>
<div style="line-height: 100%;">
<br /></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.11"></a><b>Can
I CANCEL requests other than the first INVITE?</b></span></div>
<div style="line-height: 100%;">
<br /></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Yes,
any request can be cancelled before it has been executed by the UAS.
However, it is likely that this will only make sense in practice for
the initial <var>INVITE</var>
and subsequent "re"<var>INVITE
(as for INVITE it may take longer time to get a final response than
for non-dialog-establishing type of requests, such as OPTIONS or
INFO)</var>. Btw, In
the latter case ("re"<var>INVITE</var>),
the call remains, just any changes requests are cancelled.</span></div>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.12"></a><b>What
is the relationship between the From, Contact, Via and
Record-Route/Route headers?</b></span><br />
<span style="font-family: Verdana, sans-serif;">All these headers determine
how requests and responses are routed in a network of SIP proxy
servers. Roughly, the distinction is:
</span><br />
<span style="font-family: Verdana, sans-serif;"><var>From</var>:</span><br />
<span style="font-family: Verdana, sans-serif;">Used for subsequent
requests if there is no <var><i>Contact</i></var>
or <var>Record-Route</var>
header. E.g., if Alice makes a call with From: Alice
<alice@example.org> to Bob, an <i>INVITE</i>
request from Bob to Alice would use alice@example.org as the To
header and Request-URI.
</span><br />
<span style="font-family: Verdana, sans-serif;"><var>Contact</var>:</span><br />
<span style="font-family: Verdana, sans-serif;">Determines the destination
placed in the Request-URI for subsequent requests and can be used to
bypass proxies not enumerated in a Record-Route header. Also used in
responses by redirect servers and in <var>REGISTER</var>
requests and responses.
</span><br />
<span style="font-family: Verdana, sans-serif;"><var>Record-Route</var>/<var>Route</var>:</span><br />
<span style="font-family: Verdana, sans-serif;">The Record-Route header is
inserted into requests by proxies that want to be in the path of
subsequent requests for the same call-id. It is then used by the user
agent to route subsequent requests. The mechanism is similar to a
source-route, copying the Record-Route information into a set of
Route headers. The Request-URI is set to the first Route header.
</span><br />
<span style="font-family: Verdana, sans-serif;"><var>Via</var>:</span><br />
<span style="font-family: Verdana, sans-serif;">Via headers are inserted by
servers into requests to detect loops and to allow responses to find
their way back to the client. They have no influence on the routing
of future requests (or responses).
</span><br />
<span style="font-family: Verdana, sans-serif;">Generally, in short,
requests should be sent to <var>Route</var>
if present, <var>Contact</var>
if there is no <var>Route</var>,
<var>From</var>
if there is no <var>Contact</var>.
</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.13"></a><b>How
are URLs compared?</b></span><br />
<div style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">Two
SIP URLs are compared for equality according to the following rules:
</span></div>
<table cellpadding="0" cellspacing="0" style="width: 100%px;">
<colgroup><col width="16*"></col>
<col width="240*"></col>
</colgroup><tbody>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image13" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">the
display name is ignored;
</span><br />
</td>
</tr>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image14" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">tags
must match;
</span><br />
</td>
</tr>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image15" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">user,
password, host, port and parameters of the URI must match. If a
component is omitted, it matches based on its default value.
</span><br />
</td>
</tr>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image16" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">string
comparisons are case-insensitive;
</span><br />
</td>
</tr>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image17" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">Characters
other than those in the "reserved" and "unsafe"
sets (see RFC 2396) are equivalent to their ""%"
HEX HEX" encoding.
</span><br />
</td>
</tr>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image18" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">An
IP address that is the result of a DNS lookup on a hostname does
<b>not</b> match that hostname.</span><br />
</td>
</tr>
</tbody></table>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.14"></a><b>What's
the difference between the request URIs tel:+12125551212 and
<a href="mailto:sip:12125551212@gw.com">sip:12125551212@gw.com</a>?</b></span><br />
<span style="font-family: Verdana, sans-serif;">Non-SIP URLs, such as
tel:+12125551212 for a telephone number, may be used as request URIs
in SIP <var>INVITE</var>
requests. This only makes sense if all outbound calls are handled by
a proxy server. In the case of a tel: URL, the proxy server would
then translate the request URL to a SIP URL of a gateway server, if
it is not handling the gateway duty itself. The proxy server might
use the Gateway Location Protocol (GLP) to find the appropriate
next-hop SIP server. The <var>To</var>
header may always be a tel: URL even if the Request-URI is a SIP URL,
although that breaks with the common practice that Request-URI and <var>To</var>
start out the same.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.15"></a><b>Does
SIP do admission control?</b></span><br />
<span style="font-family: Verdana, sans-serif;">Since this offers no real
security (calls could always bypass a server), admission control is
not supported by SIP. If an "outbound proxy" is used for
outgoing calls, that proxy may control the firewall and thus restrict
outgoing calls.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.16"></a><b>Does
SIP administer bandwidth?</b></span><br />
<span style="font-family: Verdana, sans-serif;">No, that is the role of a
resource reservation protocol. There is no reason to assume that any
Internet telephony signaling server (such as a proxy) would know the
available bandwidth in real networks. Having such a central server
would not scale. Administering bandwidth separately for each
application is also likely to be difficult and inefficient.
</span><br />
<span style="font-family: Verdana, sans-serif;">There is a proposal for an
SDP extension (<a href="http://www.ietf.org/rfc/rfc3312.txt">RFC
3312</a>) that allows SIP INVITE requests and responses to indicate
that resource reservation must succeed before the callee is alerted
(was initiated by 3GPP as part of IMS).
</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.17"></a><b>Do I
always need a proxy or redirect server?</b></span><br />
<span style="font-family: Verdana, sans-serif;">No, two SIP endpoints can
contact each other directly.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.18"></a><b>How
does a caller find its proxy server?</b></span><br />
<span style="font-family: Verdana, sans-serif;">Calls typically proceed
directly to the callee's domain. For example, when calling
alice@example.com the <var>INVITE</var>
request would be sent to the SIP server for the domain example.com
found via DNS.
</span><br />
<span style="font-family: Verdana, sans-serif;">If a "local"
(outbound) proxy is needed for outgoing calls, it currently needs to
be manually configured, similar to the configuration of web proxies
in browsers. Extensions to (for example) use a <var>REGISTER</var>
response or DHCP are under discussion.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.19"></a><b>What's
the difference between a stateless and a stateful proxy server?</b></span><br />
<span style="font-family: Verdana, sans-serif;">Stateless proxies forget
about the SIP request once it has been forwarded. Stateful proxies
remember the request after it has been forwarded, so they can
associate the response with some internal state. In other words,
stateful proxies maintain <i>transaction</i>
state. <em>Stateful
implies transaction state, </em><em><b>not</b></em><em>
call state.</em>
</span><br />
<span style="font-family: Verdana, sans-serif;">Stateless proxies scale
very well, and can be very fast. They are good for network cores.
Stateful proxies can do more (they can fork, for example, see the
next question) and can provide services stateless ones can't (call
forward busy, for example). They don't scale as much as stateless
ones. An administrator gets to decide which to use. These are also
logical entities; a physical proxy is likely to act as a stateless
proxy for some calls, stateful for others, and as a redirect server
for even others.
</span><br />
<span style="font-family: Verdana, sans-serif;">Neither stateful nor
stateless proxies need to maintain call state, although they can, but
will need to make sure that they are part of subsequent transactions
via the <i>Record-Route</i>
header.
</span><br />
<span style="font-family: Verdana, sans-serif;">Proxies must be stateful if
one of the following conditions hold:</span><br />
<ol>
<li><div style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">uses
TCP,
</span></div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">uses
multicast,
</span></div>
</li>
<li><span style="font-family: Verdana, sans-serif;">forks.
</span><br />
</li>
</ol>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.20"></a><b>Why
can a forking SIP proxy not be stateless?</b></span><br />
<span style="font-family: Verdana, sans-serif;">A forking SIP proxy cannot
be stateless because it needs to perform a filtering operation,
returning (in general) one response out of the many it receives. For
example, a forking proxy with three branches, that receives a
200-class, 400-class, and 500-class response on each branch
respectively, should return only the 200-class response upstream. If
the proxy were stateless, it would end up returning all three of the
responses upstream (since it won't remember that it had received
prior responses when it gets another one). The result of this is (1)
response implosion at the client, and (2) inconsistent responses at
the client. (In this example, depending on the order the responses
would be received, the client would think that the call failed, just
to get a success indication some time later.) Thus, a forking proxy
must be stateful.
</span><br />
<span style="font-family: Verdana, sans-serif;">Also note that a proxy that
uses TCP must be stateful as well, whether it forks or not. This has
to do with reliability issues.
</span><br />
<span style="font-family: Verdana, sans-serif;">Why do you want state in a
proxy? Certain services (like forking) simply require it. A
sequential search proxy requires state; sequential search is the
heart of services like follow-me and personal mobility. It's at the
discretion of the implementor whether to use a stateful or stateless
proxy. You can even be "super stateful", and use the
Record-Route header to allow a proxy to be on the signaling path of
all subsequent exchanges. This allows a stateful proxy to maintain
call state in addition to transaction state.
</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.21"></a><b>How
does a caller find the remote SIP client of the callee?</b></span><br />
<span style="font-family: Verdana, sans-serif;">The process is similar to
the delivery of email: The caller uses the SIP host name to look up
the destination host, first trying a NAPTR/SRV record and then
"regular" DNS, just like an email client (MTA) looks up the
MX record. (NAPTR/SRV records are generalized MX records applicable
to any network service, including, but not limited to, SIP and RTSP.)
For example, when contacting bell@cs.columbia.edu the client finds a
NAPTR/SRV record pointing to erlang.cs.columbia.edu as the SIP server
for the domain cs.columbia.edu. As for email, a single domain name
can resolve to multiple servers, allowing load sharing and
redundancy. (NAPTR assists in selecting the (<i>server
that supports the</i>)
desire transport, e.g. SIP over TCP)
</span><br />
<span style="font-family: Verdana, sans-serif;">The server located in this
manner can then proxy or forward the call to another server.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.22"></a><b>How
does SIP get through a firewall or NAT?</b></span><br />
<span style="font-family: Verdana, sans-serif;">There are several possible
approaches to SIP-capable firewalls. One of the difficulties is that,
unlike for, say, HTTP, connections are originated both by hosts
inside and outside the firewall. A likely arrangement is that a SIP
proxy sits "on" the firewall and relays SIP requests
between the Internet and the intranet. This proxy would also open up
the necessary ports in the firewall to let audio and video flow
through, for example using <a href="http://web.archive.org/web/20030213031908/http://www.socks.nec.com/socksv5.html">Socks
V5</a>. (Such server would normally be referred to as ALG (App. Layer
Gateway))
</span><br />
<span style="font-family: Verdana, sans-serif;">As an alternative, if a
firewall or NAT allows outgoing TCP connections, the inside client
can open up a TCP connection to an outside proxy. All outgoing and
incoming calls would then be handled by that TCP connection. (The
client would still have to use SOCKS or similar mechanism to convince
the firewall to let RTP packets through.)
</span><br />
<span style="font-family: Verdana, sans-serif;">As of 2006 the key
solutions for NAT/firewall traversal are: SBC (Session Border
Controller), ALG and using the STUN protocol (RFC 3489).</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.23"></a><b>How
does SIP do "call progress tones" or "ring back"?</b></span><br />
<span style="font-family: Verdana, sans-serif;">The
SIP server being called, such as an Internet telephony gateway, can
return any number of provisional status messages that indicate call
progress. Typically, this is just 100 (Trying) followed by 180
(Ringing), but a server could produce elaborate feedback such as
</span><br />
<pre class="western"><span style="font-family: Verdana, sans-serif;">100 Message received</span></pre>
<pre class="western"><span style="font-family: Verdana, sans-serif;">100 Looking up number</span></pre>
<pre class="western"><span style="font-family: Verdana, sans-serif;">100 Found number, looking up carrier according
to profile</span></pre>
<pre class="western"><span style="font-family: Verdana, sans-serif;">100 Finding cheapest carrier which doesn't do
animal testing</span></pre>
<pre class="western"><span style="font-family: Verdana, sans-serif;">100 Found carrier "AT&T"</span></pre>
<pre class="western"><span style="font-family: Verdana, sans-serif;">100 Dialing number</span></pre>
<pre class="western"><span style="font-family: Verdana, sans-serif;">180 Ringing</span></pre>
<pre class="western"><span style="font-family: Verdana, sans-serif;">182 Queued, 3 people in front of you</span></pre>
<pre class="western" style="margin-bottom: 0.5cm;"><span style="font-family: Verdana, sans-serif;">182
Queued, 2 people in front of you</span></pre>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">The
language of the status message should be determined based on the
<var>Accept-Language</var>
request header in the call.
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">A
183 (Session Progress) status response will appear in RFC2543bis. It
can be used for both progress tones as well as error messages.
</span></div>
<span style="font-family: Verdana, sans-serif;">One would use the 183 only
if you:</span><br />
<table cellpadding="0" cellspacing="0" style="width: 100%px;">
<colgroup><col width="16*"></col>
<col width="240*"></col>
</colgroup><tbody>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image19" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">Are
able to determine that the audio being generated is something
other than ringing (e.g. "comfort tone" or "pay
tone" as defined in E.18x), or
</span><br />
</td>
</tr>
<tr>
<td style="border: none; padding: 0cm;" width="6%">
<span style="font-family: Verdana, sans-serif;"><img align="bottom" border="0" height="15" hspace="13" name="Image20" src="data:image/png;base64,iVBORw0KGgoAAAANSUhEUgAAAA8AAAAPCAMAAAAMCGV4AAAALVBMVEUAAAAzAAAzMzNmZmZmmZmZZmb/AAD/ZmaZmZmZzMzMzMz////8A/sAAAD///86ylaYAAAADXRSTlP///////////////8APegihgAAAGZJREFUeJxVj9EOxSAMQplsysr9/++9tS5m48F4oGkofl9hvaXN4C0G8TBkX6cdKIaY/7DZkAyOPjxFM5cgjt4fJyafUho9UZVfuW4UUzXvrcr55Rzgshqx+rRcpZnuvqS4+77v+QNr5QcJ+PirtQAAAABJRU5ErkJggg==" width="15" /></span><br />
</td>
<td style="border: none; padding: 0cm;" valign="top" width="94%">
<span style="font-family: Verdana, sans-serif;">Are
unable to definitively determine that alerting is occuring. This
really should only happen with older CAS protocols. ISUP and ISDN
have sufficient information to determine what is happening on the
far end.</span><br />
</td>
</tr>
</tbody></table>
<span style="font-family: Verdana, sans-serif;">One can also use 183 if the
gateway is able to determine that an error has occured, but that
there is a tone or announcement accompanying it (e.g., an ACM with a
cause code present). In that case, the gateway can send a 183 to set
up the media for the announcement (ideally with the announcement text
as the text string), wait for a timer (on the order of 30 seconds),
and then send an appropriate SIP error message.
</span><br />
<span style="font-family: Verdana, sans-serif;">However, this should only
be done if the caller is likely a human being, as sending 183 would
otherwise only delay failure handling.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.24"></a><b>Does
SIP do keep-alive?</b></span><br />
<span style="font-family: Verdana, sans-serif;">Originally it didn't, but
now it does. See <a href="http://www.ietf.org/rfc/rfc4028.txt">Session
Timers (RFC 4028)</a>. (Thank you Vagelis Kalligeris for the
typo correction!)</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.25"></a><b>Why
does SIP not have a Content-Transfer-Encoding header?</b></span><br />
<span style="font-family: Verdana, sans-serif;">The
<var>Content-Transfer-Encoding</var>
header was primarily meant to allow message bodies to be transformed
into formats that could be transferred on channels that were not 8
bit clean. HTTP, which makes use of many of the MIME headers, is 8
bit clean, and thus did not need <var>Content-Transfer-Encoding</var>.
SIP followed suit, and so does not use it either. <var>Content-Encoding</var>
is used for things like compression, which is different. (J.
Rosenberg)
</span><br />
<span style="font-family: Verdana, sans-serif;">See also RFC 2616
(HTTP/1.1), Section 19.4.5.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.26"></a><b>I
want SIP to be more compact. What can I do? </b></span><br />
<span style="font-family: Verdana, sans-serif;">First, one should realize
that in general, SIP exchanges are only going to be a tiny fraction
of the overall session bandwidth. A typical SIP call setup takes less
than 1000 bytes, or the equivalent of one second of highly compressed
(G.729) audio. Some additional space savings can be realized by using
short headers. (IMS makes things a bit more complicated though due to
its many extensions)
</span><br />
<span style="font-family: Verdana, sans-serif;">In general, more
substantive savings are possible by using either <a href="http://www.ietf.org/html.charters/rohc-charter.html">Rohc</a>/Signaling
Compression (SigComp RFC 3320/3321/3486 etc) or IP payload
compression (<a href="http://web.archive.org/web/20030213031908/http://www.normos.org/rfc/rfc2393.txt">RFC
2393</a>) or link-layer compression, e.g., at the PPP layer. For the
example above, the total size is reduced to about 520 bytes with gzip
compression.
</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.27"></a><b>What
are the different addresses in SIP?</b></span><br />
<span style="font-family: Verdana, sans-serif;">SIP INVITE requests involve
three addresses:
</span><br />
<ol>
<li><div style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">The
host address where the request came from. Responses are sent back to
the same host address, regardless of what the <var>From</var>
header indicates. Note that different requests for the same call can
come from different hosts.
</span></div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">The
<var>From</var>
address contains the logical source of the request. It remains
unmodified as a SIP request traverses proxies, for example. The <var>From</var>
address may not be the same as the host address that generated the
SIP request, although that's the typical case.
</span></div>
</li>
<li><span style="font-family: Verdana, sans-serif;">The session
description (e.g., SDP) contains one or more addresses where the
caller expects media data (audio, video) to be sent. For some
services, this address may not be the same as the <var>From</var>
address.
</span><br />
</li>
</ol>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.28"></a><b>How
do I put call on hold?</b>
</span><br />
<span style="font-family: Verdana, sans-serif;">There are several
"traditional" ways to do that, e.g. zeroing the IP address
or port number in the media descriptor of the stream to be placed on
hold. However the most correct and up-to-date way is to set media
attribute parameter to 'inactive or sendonly', i.e. "a=inactive"
or "a=sendonly".</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q2.29"></a><b>In
what practical scenarios Call-Info header is(/can be) used?</b></span><br />
<span style="font-family: Verdana, sans-serif;">The Call-Info header field is included in a request by a UAC or
proxy to provide a URI with information relating to the session
setup. It may be present</span><br />
<span style="font-family: Verdana, sans-serif;">in an INVITE, OPTIONS or REGISTER request. The header field
parameter purpose indicates the purpose of the URI and may have the
values</span><br />
<span style="font-family: Verdana, sans-serif;">icon, info, card, or other IANA registered tokens. An example
follows:</span><br />
<span style="font-family: Verdana, sans-serif;">Call-Info:
<<a href="http://www.code.com/my_picture.jpg"><span style="color: blue;"><u>http://www.code.com/my_picture.jpg</u></span></a>>;purpose=icon.
(Karimulla Sayed)</span><br />
<br />
<table bgcolor="#808000" cellpadding="2" cellspacing="0" style="width: 415px;">
<colgroup><col width="405"></col>
</colgroup><tbody>
<tr>
<td style="border: 2.25pt solid #000000; padding: 0.05cm;" width="405">
<div align="center">
<b><span style="font-family: Verdana, sans-serif;"><span style="color: white;"><span style="background: #808000;">---
Relationship To Other Protocols</span></span><span style="color: white;"><span style="background: #808000;">
</span></span><span style="color: white;"><span style="background: #808000;">---</span></span></span></b></div>
</td>
</tr>
</tbody></table>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.1"></a><b>Does
SIP do conference control?</b></span><br />
<span style="font-family: Verdana, sans-serif;">SIP leaves conference
control, such as the election of a chair or floor control, to other
protocols. SIP can be used for non-conferencing applications and
floor control may be used outside the scope of SIP-initiated calls,
so it seemed best to separate the functionality. However, SDP may be
used to indicate which media are subject to floor control and what
tools and protocols are to be used. This is work in progress mainly
in the <a href="http://www.ietf.org/html.charters/xcon-charter.html">IETF</a>
and <a href="http://www.sipknowledge.com/www.openmobilealliance.org">OMA</a>
standardization bodies.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.2"></a><b>What
is the relationship between MGCP/Megaco/H.GCP and SIP?</b></span><br />
<span style="font-family: Verdana, sans-serif;">The details of combining
the two in a system are still being fleshed out. MGCP is a <em>device
control protocol</em>,
where a slave (gateway (MG)) is controlled by a master (media gateway
controller (MGC), call agent). SIP may be used <em>between</em>
controllers, in a peer-to-peer relationship. Note that to the SIP
side, the MGC looks like a node with a large number of connections,
but otherwise the same as a "native" SIP device. Similarly,
the MG is completely unaware that the call between MGCs is
established via SIP. Only the MGC needs to understand both protocols.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.3"></a><b>What
is SIP+ and how does it relate to SIP?</b></span><br />
<span style="font-family: Verdana, sans-serif;">SIP+ was a proposal by
Level3 on how to extend SIP to interconnect two MGCs. This
functionality is now being provided by various orthogonal SIP
extensions, including the carriage of multipart MIME types, the INFO
method and others. These are being documented in a BCP draft. The
name SIP+ is obsolete and should not be used to avoid confusion.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.4"></a><b>How
does SIP compare to H.323?</b></span><br />
<span style="font-family: Verdana, sans-serif;">The H.323 protocol came on
the scene in the mid-'90s as a transmission and session setup
protocol for videoconferencing over ISDN networks. It comes out of
the International Telecommunication Union (ITU), a 54-year-old
standards body for technologies and protocols for the international
phone network. <br />H.323 is not a single protocol in one vertical
integrated stack, but it is a suite of protocols that cover codecs,
call control, conferencing, and many other functions. <br />The
advantage to this approach is that by strictly controlling so many
aspects of the implementation it is easier to ensure that H.323 based
systems function well together. <br />On the down side, H.323 has
become heavy and cumbersome. Flexibility is sacrificed as one is tied
to a single family of technologies. For a field as young and fast
changing as IP telephony, where many problems and solutions are still
under debate, flexibility is an important aspect. <br />SIP is part of
this flexible approach, as it uses a wide variety of protocols, each
addressing a different aspect of the problem space. The advantage is
the ability to choose from among many competing technologies and move
to newer and better ones as they emerge. This has always been the
philosophy behind SIP and this is the approach of the IETF to IP
telephony in general. (Taken from SIP Illustrated)
</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.5"></a><b>Can
SIP be used for Internet telephony gateways (ITGs)?</b></span><br />
<span style="font-family: Verdana, sans-serif;">Yes, in two ways. First, it
can indicate to the Internet-based caller that the callee is
reachable via an ITG, via the <var>Contact</var>
header. Secondly, two ITGs connecting parties on the PSTN can signal
new calls to each other, with the destination phone number contained
in the request URL.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.6"></a><b>Can
H.323 and SIP be used together?</b></span><br />
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Yes.
SIP can locate the called party and determine its capabilities,
including H.323. H.323 is then used to connect the two parties.
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">Unfortunately,
there is currently no specification on translating between the two.
Conversion is made more difficult by the multiple versions of H.323
(v1, v2, v3). However, there are several gateway products in the
market place that allows SIP and H.323 terminals to call each other.
</span></div>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.7"></a><b>How
do I interconnect Q.931 (ISDN signaling) and SIP?</b></span><br />
<span style="font-family: Verdana, sans-serif;">A gateway that initiates an
ISDN call based on a SIP call or vice versa is reasonably
straightforward.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.8"></a><b>How
do I interconnect ISUP (SS7 signaling) and SIP?</b></span><br />
<span style="font-family: Verdana, sans-serif;">Similar to the above. <a href="http://www.sipknowledge.com/SIP_RFC.htm">SIP-T
and SIGTRAN</a> provide standardization in this area.</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.9"></a><b>What
is sip-cgi and how does it relate to CPL?</b>
</span><br />
<span style="font-family: Verdana, sans-serif;">Both are viewed as
different approaches for creating VoIP services. Both are written
offline, and both are executed when messages arrive in order to
execute features.</span><br />
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">CPL
is an XML-based language, while sip-cgi is a mechanism for invoking
scripts or programs written in any language. sip-cgi is very similar
to web cgi scripts.
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;">In
its current version, CPL is only invoked when <var>INVITE</var>
requests and responses arrive, while sip-cgi can intercept any
request.
</span></div>
<span style="font-family: Verdana, sans-serif;">sip-cgi is designed to be
used by SIP, while CPL can probably be used by a number of signaling
protocols such as Q.931 or H.323.
</span><br />
<span style="font-family: Verdana, sans-serif;">CPL and sip-cgi differ in
their applicability. CPL is designed for end user service creation.
It is intentionally limited in capabilities and is not a general
purpose programming language. Its execution on a server is generally
very fast. CGI is more powerful - you can do nearly anything. It is
programming language independent. It incurs a process-spawning
overhead, so its less efficient than CPL. (CPL is usually executed in
the same process as the server). As a service provider, I would not
want to execute CGI scripts sent to me by end users. However, I would
prefer to use CGI to develop my own services.
</span><br />
<span style="font-family: Verdana, sans-serif;">Note that CGI may be used
as the execution environment for a CPL script. (Jonathan Rosenberg)
</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q3.10"></a><b>Is
there a SIP interoperability certification? How can I test
interoperability with others?</b></span><br />
<span style="font-family: Verdana, sans-serif;">Check out the <a href="http://www.sipit.net/sipit.php">SIPit</a>
events (IOT) and the <a href="http://www.sipforum.org/">SIP Forum</a>
(certification).</span><br />
<table bgcolor="#808000" cellpadding="2" cellspacing="0" style="width: 415px;">
<colgroup><col width="405"></col>
</colgroup><tbody>
<tr>
<td style="border: 2.25pt solid #000000; padding: 0.05cm;" width="405">
<div align="center">
<b><span style="font-family: Verdana, sans-serif;"><span style="color: white;"><span style="background: #808000;">---
SIP Implementation/tools</span></span><span style="color: white;"><span style="background: #808000;">
</span></span><span style="color: white;"><span style="background: #808000;">---</span></span></span></b></div>
</td>
</tr>
</tbody></table>
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q4.1"></a><b>Is
there any free testing tool, which can build SIP messages and upon
reception can respond with
specific predefined response?</b></span><br />
<span style="font-family: Verdana, sans-serif;">SIPp (this is not a
typo...) (<a href="http://sipp.sourceforge.net/"><span style="color: blue;"><u>http://sipp.sourceforge.net</u></span></a>)
is a fantastic tool for accomplishing what I think you're after.
[Rhys Ulerich]</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q4.2"></a><b>Is
there any tool that generates SIP log (call trace)?</b></span><br />
<span style="font-family: Verdana, sans-serif;">Yes. check out Call
Analyzers from: Ethereal (<a href="http://www.ethereal.com/">www.ethereal.com</a>),
Empirix (<a href="http://www.empirix.com/">www.empirix.com</a>),
QuadTex (<a href="http://www.quadtexsys.com/">www.quadtexsys.com</a>).</span><br />
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q4.3"></a><b>Are
there any tools that would allow generation of </b><u><b>graphical</b></u><b>
SIP</b> <b>call</b>
<b>flows?</b></span><br />
<span style="font-family: Verdana, sans-serif;">You might want to look at
SIP Scenario: <a href="http://www.iptel.org/~sipsc/"><span style="color: blue;"><u>http://www.iptel.org/~sipsc/</u></span></a>
It does a few things that the
Ethereal analyser doesn't, but takes a little more effort to
configure (Michael Procter). Also Ethereal does this itself after the
10.2 version.</span><br />
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;"><a href="https://www.blogger.com/null" name="q4.4"></a><b>Should
a terminal, which is intended to transmit calls to the PSTN, support
TRIP (Telephony Routing Over IP) or is it only the problem </b>
</span></div>
<div style="line-height: 100%;">
<span style="font-family: Verdana, sans-serif;"><b>of
the gateway?</b></span></div>
<span style="font-family: Verdana, sans-serif;">TRIP is not needed in end
user phones or PC clients. It is used between servers facing each
other in a peering relationship between service<br />providers. Usage
scenarios for TRIP are described fully in:
<a href="http://www.ietf.org/rfc/rfc2871.txt">http://www.ietf.org/rfc/rfc2871.txt</a><br /><br />A
client or PC phone which wishes to make a call to the PSTN can do one
of several things. Starting with the most commonly implemented
one:<br /><br />1. If the phone is in domain foo.com, it constructs a SIP
URL of the form sip:<dialed-number>@foo.com, puts that in the
request URI, and sends the request. The server for its domain figures
out what to do, using things like ENUM, TRIP, or statically
configured routing tables.<br /><br />2. The phone inserts a tel URL into
the request URI, of the form tel:<dialed-number>, and sends it
to its proxy. From there, things proceed as above in step 1.<br /><br />3.
The phone uses ENUM itself, and possibly gets back a SIP URL for that
number, which it can use directly. If the ENUM query fails to yield a
SIP URL (which will be a frequent occurence), proceed to step 1 or
2.<br /><br />(-Jonathan R.)</span><br />
<br />
<span style="font-family: Verdana, sans-serif;"><br /></span><br />
</div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com2tag:blogger.com,1999:blog-5206096933420493644.post-16949577463359350602015-04-29T01:01:00.002-07:002015-04-29T01:01:43.565-07:00Relations between IP, Basic Telephony, 3GPP/3GPP2, PTT, PoC, IMS and SIP<div dir="ltr" style="text-align: left;" trbidi="on">
<h2 class="western">
<span style="font-size: small;">1.1<span style="color: black;"><span style="font-family: Arial;"><b>How
are IP routers related to SIP servers? Are IP routers SIP aware?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">SIP
servers are plain hosts on the Internet or the intranet. They do not
do IP routing. they only forward SIP messages just like email agents
do with email messages (SMTP). IP routers are NOT SIP aware. Normally
they do not look at IP packets beyond the IP header bits, thus they
have no idea whether the payload is SIP, HTTP, SMTP, DNS, Telnet, FTP
or something else.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q2"></a>1.2<span style="color: black;"><span style="font-family: Arial;"><b>Any
relation between SIP URL and IP address?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Not
really. SIP URL is an application layer Identifier, similar to an
email or web address. It is associated with a person not a computer.
IP address is associated with a computer (see the IP Basics section
above). Hence a computer may host multiple SIP URLs (even at the same
time). In addition, the same SIP URL might be hosted by different
computers (even) at the same time). The association between SIP URL
and IP address (of a SIP phone) is done at the SIP Registrar by a
process known as SIP Registration.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q3"></a>1.3<span style="color: black;"><span style="font-family: Arial;"><b>Can
SIP work with IPV6?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Yes,
as long as the device that routes the SIP messages support IPV6.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q4"></a>1.4<span style="color: black;"><span style="font-family: Arial;"><b>How
does SIP differ from Mobile IP (MIP)?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">SIP
phone needs to know the IP address of the next SIP hop in order to be
able to forward/send it a SIP message. However, it doesn't know
and/or doesn't care whether the next hop stays all the time at the
same place or whether it moves, as long as the IP packet, which carry
the SIP message can reach it. should the next hop move (and thus
possibly change its point of attachment to the Internet) MIP can take
care of the physical delivery of the IP packets destined to it.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q5"></a>1.5<span style="color: black;"><span style="font-family: Arial;"><b>If
a SIP UAS is unreachable, will the UAC get a SIP error code or an
ICMP error?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">It
is a bit like mixing apples and vegetables... For instance, a device
might be IP/ICMP reachable, and even SIP reachable, and yet the SIP
URL might be wrong or not handled by the destination SIP device, and
thus a SIP "404 Not Found" might be returned. In a
different situation the SIP URL might be valid, but the target host
has crashed and thus the router at the destination network may return
ICMP error code 1 (Host Unreachable). Needless to say, in a situation
like that there is no target SIP application alive that can send any
SIP error back...</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q6"></a>1.6<b>Can SIP messages get
fragmented by IP nodes?</b>
</span></h2>
<span style="font-family: Arial;"><span style="color: black;">In
theory yes. However the SIP standard tries to prevent this situation
by recommending the use of congestion controlled transport protocol,
(such as TCP) in case a request is within 200 bytes of the path MTU
or larger than 1300 bytes (and the path MTU is unknown). The
motivation for preventing fragmentation of SIP messages is having
practical issues with some of the real time IP stacks.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q7"></a>1.7<span style="color: black;"><span style="font-family: Arial;"><b>Is
it a must for SIP to work with IP?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">In
theory no, but practically that's always the case.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q8"></a>1.8<b>What are codecs? What's
the relation between them and SIP?</b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Codecs
are hardware/software means that are used to encode analog
audio/video signals in binary digital format. Codecs are mainly
different from each other by the sampling rate (~bandwidth) and
number of bits they use to encode the signal samples. In VoIP they
are normally used as the payload of RTP, e.g. G711/RTP, H261/RTP etc.
SIP does not use codecs, but as part of in the call setup SIP end
points indicate (via SDP/SIP) what codecs they will use in the
conversation phase (RTP).</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q9"></a>1.9<b>Do 3GPP/3GPP2 use SIP?</b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Yes
they do. 3GPP adopted SIP in 2001. Your humble slaves were involved
in the discussions that preceded that critical decision (i.e.
preferring it on H323). 3GPP came up with the IMS (IP Multimedia
subsystem), which is their way to implement VoIP services over GPRS
and UMTS cellular networks. Defacto IMS can support any access
network technology including Wi-Fi, CDMA and others. 3GPP2 (CDMA 3rd
generation) are in a catch up mode. They are quickly adopting the
3GPP IMS design. There are some nuances between the 3GPP IMS and the
3GPP2 IMS, but these are not major. The 3GPP2 folks call their IMS -
MMD (Multimedia Domain) or All IP core.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q10"></a>1.10<span style="color: black;"><span style="font-family: Arial;"><b>What
is PTT? What is PoC? How are these related to SIP?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">PTT
stands for Push To Talk. This is the telephony technology which
simulates walkie talkie type of communication. It has become very
popular in the wireless arena. Typically when people hear this term
they say "Oh, the Nextel phones...". PoC stands for PTT
over Cellular. The OMA standard organization defines how PoC should
work. They base their work on SIP (and RTP/RTCP).</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q11"></a>1.11<span style="color: black;"><span style="font-family: Arial;"><b>How
do you do billing for SIP? PTT? PoC? IMS?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Diameter
(IETF RFC 3588) seems to be the protocol of choice for doing billing
(charging). It was adopted by 3GPP. Most likely the closely related
standard organizations will follow suit.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q12"></a>1.12<span style="color: black;"><span style="font-family: Arial;"><b>How
do you do lawful intercept for those...?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Standard
takes care of this too. The main idea is to intercept and report to
the authorities both events and data, and do so in a common way for
all VoIP related technologies/systems.</span></span><br />
<br />
<br /><br />
</div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com2tag:blogger.com,1999:blog-5206096933420493644.post-25243292788695269752015-04-29T00:59:00.002-07:002015-04-29T00:59:16.195-07:00Basic wireless/3GPP<div dir="ltr" style="text-align: left;" trbidi="on">
<h2 class="western">
<span style="font-size: small;">1.1<b>What’s the difference
between GPRS and UMTS?</b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">This
is quite confusing for many people. According the 3GPP standard (TS
23.060) GPRS defines the packet data services provided by a common
core network to two main types of access networks - GSM based
and WCDMA based (UMTS).</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q2"></a>1.2<b>What’s
the relation between GPRS and IP?</b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">GPRS
uses IP based signaling/control protocols between its components.
Beyond that it transparently carries IP packets between the mobiles
and the Internet. This is done by method known as encapsulation and
tunneling.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q3"></a>1.3Can you do
voice calls with GPRS?
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">GPRS
provides an IP data pipe. Therefore by definition you can do anything
on top of it including voice. QoS is a major issue though for voice
over GPRS, and IMS (IP Multimedia Subsystem) tries to resolve it.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q4"></a>1.4What is the
relation between GPRS and SIP?
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">SIP
rides transparently over GPRS just like any other IP based protocol.
IMS and QoS muddy this picture a bit. You can read all about it in 3G
IMS Illustrated.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q5"></a>1.5What is the
relation between UMTS and SIP? </span></h2>
<span style="color: black;"><span style="font-family: Arial;">Same
as GPRS.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q6"></a>1.6What is the
relation between CDMA (1X) and SIP?
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Very
similar to GPRS/UMTS and SIP. the designers of CDMA 1X (3GPP2) has
adopted the 3G way (IMS) for doing VoIP/multimedia.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q7"></a>1.7Does SIP care
at all about the access network that uses it?
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">No,
apart from QoS impact, which may dictate what codecs will get used
and what quality you may expects from a multimedia session.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q8"></a>1.8Can SIP run on
top of 802.11?
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Sure.
In SIP's eyes 802.11 is just another type of access network, which
can carry it.</span></span><br />
<br />
<br /><br />
</div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com1tag:blogger.com,1999:blog-5206096933420493644.post-82500578006173342622015-04-29T00:56:00.001-07:002015-04-29T00:56:33.064-07:00VoIP- Telephony Basics<div dir="ltr" style="text-align: left;" trbidi="on">
<h2 class="western">
<span style="font-size: small;">
1.1<span style="color: black;"><span style="font-family: Arial;"><b>What
is Trunk circuit? Do telephony switches have addresses?</b></span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Trunk
circuit is a single time slot, which is used to carry voice bits
(PCM) on a trunk (or span or channel) between two switches. Telephony
switches are addressable by Point Codes (The SS7 equivalence of IP
address).</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q2"></a>1.2<span style="color: black;"><span style="font-family: Arial;"><b>What
is SS7?</b></span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">SS7
is the classic out-of-band signaling protocol for circuit switched.
Mostly it does basic telephony signaling (ISUP), Intelligent network
signaling (TCAP) and cellular signaling (MAP). </span></span>
<br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q3"></a>1.3<span style="color: black;"><span style="font-family: Arial;"><b>What
is bearer? What’s the difference between bearer and media?</b></span></span>
</span></h2>
<span style="color: blue;">'</span><span style="color: black;"><span style="font-family: Arial;">Bearer'
is the circuit way of saying 'media'. In the circuit days media used
to be mainly voice (and fax, data and video to a certain extent). In
Internet Telephony the term media has much wider implication. It can
be anything from white board images to real time text (including of
course all legacy types of media).</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q4"></a>1.4<span style="color: black;"><span style="font-family: Arial;"><b>What
is PCM? TDM?</b></span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">PCM
stands for Pulse Code Modulation. TDM stands for Time Division
Multiplexing. Both pertain to the way analog signals, such as voice
are sampled, encoded as digital bits, and transmitted over the
digital wire (or wireless in some cases...). PCM/TDM is the main way
circuit switches use to convey the bearer to each other.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q5"></a>1.5<span style="color: black;"><span style="font-family: Arial;"><b>What
is bandwidth? What is bit rate? What are codecs? any relation between
those terms?</b></span></span><span style="color: black;"><span style="font-family: Arial;"> </span></span>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Explanation
taken from SIP Illustrated: Bandwidth and audio or video codecs are
few of those buzz words, that everyone uses, but NOT too many people
seem to know what they really mean…The lengthy explanation which
follows, will try to reveal this mystery for you, but you will need
to be brave and patient though…In order for any Voice/Media Over IP
system to function, the process of Analog to Digital (A/D) conversion
needs to take place (for instance, once the SIP signaling has setup a
media session, your SIP phone needs to convert your voice/image to IP
packets (e.g. G729/RTP/UDP/IP (voice); H261/RTP/UDP/IP (image)),
which then need to be transmitted to the other SIP phone, and
converted back by the other end point to audio/video information.
Voice, audio and video are all sources of analog signal waveforms
(i.e., Information, which is continuous in time and value), which
need to be converted to digital information (discrete in time and
value), so they can be transmitted as IP packets. The process of
converting an analog signal to a digital representation (A/D) is
consisted of three main phases:</span></span><span style="font-family: Arial;">
</span><span style="color: black;"><span style="font-family: Arial;">1.
Sampling (discretize time)</span></span><span style="font-family: Arial;">
</span><span style="color: black;"><span style="font-family: Arial;">2.
Quantizing(discretize value)</span></span><span style="font-family: Arial;">
</span><span style="color: black;"><span style="font-family: Arial;">3.
Encoding (digitize value).The inverse process of A/D, which is not
surprisingly called the D/A process, is consisted of decoding the
digital bits into the corresponding analog values and smoothing the
rectangular waveform by applying a low pass filter width width Fmax.
A device that can both encode an analog signal into a digital
representation and decode a digital signal into its equivalent analog
signal is called a coder-decoder, or codec for short. (Remember?
codecs are these things, which are advertised by the SDP element in
an SIP INVITE requests, for instance, “a=rtpmap:0 PCMU/8000”).In
order to understand what bandwidth is, we need to realize that based
on Fourier theorem any periodic function of a wave form (signal) is
made up of sine waves each contribute one frequency component to the
signal. The spectrum of these frequencies (to be precise. Fmax -
Fmin) is called bandwidth. In the case of “lowpass” signal, i.e.,
message signals such as data signal, voice, audio or video, bandwidth
is approximately equal to Fmax, since Fmin is negligible.This will
help us to understand the connection between bandwidth and bit rate
in digital systems. Now let’s indeed examine the relation between
bandwidth and bit rate in digital systems. In the binary encoding
phase of the A/D process, each of the binary digits, which is
generated, needs to be transmitted to its digital destination by
means of digital pulse. Since we have reciprocal relationship between
wave width (period duration) and its frequency, we can say that the
smaller the pulse width is - the higher its frequency, and thus the
larger its bandwidth (F=Fmax=bandwidth). Since a digital signal is
made up of multiple signal pulses of the same duration (and thus same
frequency), its bandwidth is the same as the bandwidth of a single
signal pulse (reminder: Bandwidth is the range of frequencies and for
practical purpose may be approximated by the max frequency element.
Therefore the bandwidth of a digital signal is actually determined by
the signal pulse it uses (pulse width and shape). Since one pulse is
used for one bit, the pulse width is proportional to the bit duration
and inversely proportional to the bit rate. Therefore the conclusion
is that in digital systems the bandwidth and bit rate are
proportional. The general expression, which links bandwidth and bit
rate in digital system is given by B = a*R when B is the bandwidth
required for relaying a digital signal with a bit rate of R bits per
second and 0.5 < a < 1 (depends on the pulse design). Once we
digest all of the above, we can realize that codecs would mainly
differ from each other by their sampling rate and step size. (To be
honest, there are other differences as well (e.g., PCM encoding
versus ADPCM encoding, different companding functions, linear
quantization versus non-linear quantization, compression!), but we
chose not say much about them, in order not to overwhelm you...). The
higher the sampling rate and the smaller the step size the more
accurate information we get, but the higher the number of steps (per
second) is, i.e., the larger the number of quantums (symbols) to be
encoded, and thus the longer the code words, i.e., large N, and thus
the higher the bit rate, which means higher (PCM) bandwidth
requirements. That means there is always a trade off between quality
(accuracy) and bandwidth requirements. The more quality you want –
the more bandwidth resources you have to throw in (pay!).</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q6"></a>1.6<b>Are call setup and call
establishment the same? What are they for?</b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Yes,
these are just two synonyms of the same thing. It is needed for
making sure the other end point is there and is willing to take the
call. In addition it is needed for the telephony network to allocate
all necessary resources for the call, make other necessary
preparations (e.g. open billing records, open stat records, prepare
lawful intercept hooks) and potentially run call setup time services
(e.g. translation, call forwarding). In the VoIP world call setup
also enables the end points exchange information about their media
capabilities (e.g. codecs which can be supported) and media
properties (e.g. IP address for media exchange, port number,
protocol).</span></span><br />
<br />
<br />
<br /></div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com1tag:blogger.com,1999:blog-5206096933420493644.post-50978674083452361142015-04-29T00:54:00.001-07:002015-04-29T00:54:27.798-07:00VoIP- IP Basics<div dir="ltr" style="text-align: left;" trbidi="on">
<h2 class="western">
<span style="font-size: small;">1.1<span style="color: black;"><span style="font-family: Arial;"><b>How
does IP router deliver IP packets to an IP host?</b></span></span><b>
</b>
</span></h2>
<span style="color: black;">First
we need to realize that IP router is a computer (host), which has
multiple network adapters, each associated with a different IP
address. These network adapters enable the router to be a physical
member of multiple subnets (e.g. LANs), thus enabling it to forward
IP packets from one subnet to another. You can </span><span style="color: black;"><span style="font-family: Arial;">think
of a router as an electronic version of an octopus, each one of its
legs is connected ("belongs") to a different subnet... Upon</span></span><span style="color: black;">
receipt of an IP packet, the </span><span style="color: black;"><span style="font-family: Arial;">IP
router takes a look at the IP portion (header) of the packet
(datagram). In particular it examines the destination IP address
field in the IP header. It looks up its routing table (using the
destination IP address as a key) and finds the closets matching
entry. This entry tells the router what is the next hop to go to, and
via which network adapter (AKA interface) to do so. The next hop
could be as simple as a host on the neighbor subnet, which its IP
address is the one indicated in the IP packet (and thus is the final
destination of the packet), or it could be another router that needs
to take the packet further down the road.</span></span><span style="color: black;">
In order to forward the packet to the next hop, the router needs to
</span><span style="color: black;"><span style="font-family: Arial;">discover the MAC
(Media Access Control) address of it by using ARP (see below). Then
it lets its MAC software and hardware (e.g. Ethernet module and
Ethernet network adapter) take care of the physical encapsulation and
delivery of the packet to the MAC hardware/software of the
destination (e.g. the Ethernet network adapter of the next hop). Note
that neighbor routers ALWAYS share at least one subnet (so they can
physically/directly forward packets to each other). </span></span>
<br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q2"></a>1.2<span style="color: black;"><span style="font-family: Arial;"><b>What’s
the connection/relation between the Internet and IP? </b></span></span><span style="color: black;"><span style="font-family: Arial;">
</span></span>
</span></h2>
<span style="font-family: Arial;"><span style="color: black;">IP
stands for </span><span style="color: blue;">I</span><span style="color: black;">nternet
</span>P<span style="color: black;">rotocol. It is the basic protocol (set
of rules; language) that ALL computers on the Internet (must) use in
order to speak to each other. For instance, some computers on the
Internet might be capable of sending and receiving emails; Some
others might only be capable of downloading web pages. Still all of
them MUST be capable of speaking IP. Recall that email messages and
web pages are always encapsulated in IP packets, just like letters
are 'encapsulated' in postal envelops. Same for SIP messages...</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q3"></a>1.3<b>What’s
ARP? </b>
</span></h2>
<span style="font-family: Arial;"><span style="color: black;">ARP
stands for </span><span style="color: blue;">A</span><span style="color: black;">ddress
</span><span style="color: blue;">R</span><span style="color: black;">esolution
</span><span style="color: blue;">P</span><span style="color: black;">rotocol.
It is the protocol/process used to map IP addresses to MAC (Media
Access Control) addresses. When hosts want to communicate with each
other on the same segment of network cable (subnet) they need to know
the physical addresses (MAC addresses) of each other. To do so they
either broadcast ARP queries on the network segment or use ARP cache.
The input of the ARP query/cache is the IP address of the destination
host. The output is the MAC addresses of it.</span></span><br />
<h2 class="western">
<span style="font-size: small;">1.4<b>What’s the relation
between IP and Ethernet? How are IP datagrams and Ethernet frames
different from each other?</b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">IP
packets (datagrams) are encapsulated in Ethernet frames. Computers
(hosts) that share network segment (IP subnet) can talk to each other
through their network interfaces (adapters). Every network adapter in
the world has its own unique Ethernet (MAC) address. The logic built
into it enables it to accept only frames whose destination MAC
address match its own MAC address (except for broadcast frames which
are always accepted). Perhaps now is the right time to realize that
when a web client sends a request to a web server for a web page, the
(HTTP) request is encapsulated in an IP packet. This IP packet may
travel many routers until it finally reaches its destination (the web
server). Every leap on its journey is consisted of ARP operation, MAC
(e.g. Ethernet) encapsulation and MAC delivery between two neighbor
network adapters (e.g. the adapters of two routers along the routing
path, or the adapters of the web client and the next hop router).
When a network adapter receives and accepts an Ethernet frame it
throws away the bits that constitute the Ethernet (MAC) header and
hands the remaining bits (IP header and its payload) up the stack to
the IP module. This is called decapsulation and stack propagation.</span></span><br />
<h2 class="western">
<span style="font-size: small;">1.5<b>Is TCP reliable? Is UDP
reliable?</b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Yes,
TCP is a reliable transport protocol. Its reliability is achieved by
means of acknowledgements and segments retransmission. UDP is not
reliable, but is very simple and is a good fit for packets that
contain real time media information, such as voice or video. Loosing
a single voice frame might not be that critical, but having to delay
the transmission of the next voice frame due to lack of
acknowledgement for its predecessor frame, might be critical and thus
makes TCP a bad choice for real time communication.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q6"></a>1.6<b>What is
UDP?</b>
</span></h2>
<span style="font-family: Arial;"><span style="color: black;">UDP
stands for </span><span style="color: blue;">U</span><span style="color: black;">ser
</span><span style="color: blue;">D</span><span style="color: black;">atagram
</span><span style="color: blue;">P</span><span style="color: black;">rotocol.
It is the simple standardize way to encapsulate a message and
identify its application layer destination (by using port #). UDP
header contains the source and target ports, message length and
optional checksum.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q7"></a>1.7<b>What is
TLS?</b>
</span></h2>
<span style="font-family: Arial;"><span style="color: black;">TLS
stands for </span><span style="color: blue;">T</span><span style="color: black;">ransport
</span><span style="color: blue;">L</span><span style="color: black;">ayer
</span><span style="color: blue;">S</span><span style="color: black;">ecurity.
It is a mechanism that provides hop to hop transport-layer security
over connection-oriented protocols (e.g. TCP); It provides data
integrity (no one has tampered with it), data confidentiality (no one
has seen it or was able to understand it) and authentication (i.e.,
the sender is who he claims he is).</span> <span style="color: black;">TLS
starts with a handshake phase that negotiates an encryption algorithm
(e.g., AES, IDEA) and keys, and authenticates the server to the
client (and vice versa) using certificates and trusted known CAs
(Certification Authorities). Once the handshake is complete and data
transmission begins, the data is encrypted using the keys and
algorithm negotiated during the handshake phase (Symmetric
cryptography).</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q8"></a>1.8<b>What is
port number? Why is it called port?</b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">Port
number is a logical identifier for a sender or receiver application.
It has nothing to do with the physical port devices/connectors at the
back of the computer... It enables multiplexing of IP packets between
different applications sharing the same platform. In other words: A
computer can run several applications at the same time. Each one of
these may send/receive IP packets to/from different or identical
sources. Each one of the incoming IP packets will find its way (up in
the IP stack) to the correct waiting application based on the port
number associated with it. The port number is added to a message by
the application layer and is part of the transport header (e.g. UDP,
TCP).</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q9"></a>1.9<b>What is an
IP Switch? Is it the same as telephony switch?</b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">IP
Switch is a new marketing buzz word. Normally it is used to describe
an IP telephony server. Another word which is used interchangeably
with the word 'IP switch' is Soft Switch. The IP switch (or Soft
Switch) is different from the legacy telephony switch by the strict
separation it keeps between the application logic (controller) unit
and the media switching unit. This is in contrast to the centric way
switching is done by circuit telephony switches.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q10"></a>1.10<b>What is
DNS? Any relation between Internet domains and IP subnets? </b>
</span></h2>
<span style="color: black;"><span style="font-family: Arial;">DNS
Stands for Domain Name System. It enables us to use names which are
easy to remember (e.g. <a href="http://www.nba.com/">www.nba.com</a>)
rather than long and meaningless IP addresses (e.g. 10.32.111.2).
Internet domain may contain many different IP subnets. Domain is a
logical term, and its geographical meaning might be loose in many
cases. For instance the domain 'intel.com' describes a virtual
network that is consisted of many subnets in different geographical
locations.</span></span><br />
<h2 class="western">
<span style="font-size: small;"><a href="https://www.blogger.com/null" name="q11"></a>1.11<b>What is
IP multicast? Do I need to add a network adapter to my PC to be able
to do multicast?</b>
</span></h2>
By Bryan McLaughlin -
Cisco: IP multicast is the ability for a host and the network to
enable delivery to a group of interested receivers. A host can
dynamically choose to listen/join a multicast group (using the same
network adapter). IP routers that support multicasting, take care of
duplicating the original IP packet and delivering the copies to all
listeners to the group. In short IP multicast enables an unlimited
number of hosts to receive a single data stream with 'no additional'
load on the source or the network<br />
<br />
<br /><br />
</div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com0tag:blogger.com,1999:blog-5206096933420493644.post-76739712526745281792015-04-29T00:49:00.003-07:002015-04-29T00:49:54.294-07:00IMS Technology<div dir="ltr" style="text-align: left;" trbidi="on">
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><b>IMS</b>
enables a packet -based Network to provide multiple services on
single control/service layers via different access networks.</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">IMS
Requirements:</span></b></div>
<div align="left">
<span style="font-family: Verdana, sans-serif;"><br /><br />
</span></div>
<ol>
<li value="1"><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">IP
Multimedia sessions
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">QOS
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">Service
control
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">Roaming
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">Internetworking
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">Rapid
service creater
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">Multiple
access
</span></div>
</li>
</ol>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">IMS
Protocols:</span></b></div>
<div align="left">
<span style="font-family: Verdana, sans-serif;"><br /><br />
</span></div>
<ol>
<li value="1"><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">SCF(Session
Control Function)
</span></div>
</li>
</ol>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
Cirtuit Switched network :-
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
TUP (Telephony User Part)</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
ISUP(ISDN User Part)</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
BICC(Bearer independent call
control)</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
Packet Switched Network:-</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
SIP</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
H.323</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"> 2)
AAA(Authentication, Authorization and Accounting)</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
Diameter</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"> 3)
Other protocols</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
Megaco(H248)</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
RTP/RTCP</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">IMS
was originally standardized by 3gpp.</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">Key
components or <b>Nodes of IMS</b> architecture:</span></div>
<div align="left">
<span style="font-family: Verdana, sans-serif;"><br /><br />
</span></div>
<ol>
<li value="1"><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">HSS
and SLF
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">CSCF
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">AS
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">BGCF
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">Media
Gateways
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">MRF</span></div>
</li>
</ol>
<ol>
<li value="1"><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><b>HSS
& SLF : </b>
</span></div>
</li>
</ol>
<div align="left" style="margin-bottom: 0cm; margin-left: 0.95cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div style="margin-bottom: 0cm; text-align: left;">
<b><span style="font-family: Verdana, sans-serif;">HSS(Home
Subscriber System)========:</span></b></div>
<div style="margin-bottom: 0cm; text-align: left;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div style="margin-bottom: 0cm; text-align: left;">
</div>
<ul style="text-align: left;">
<li><span style="font-family: Verdana, sans-serif;">It
is a database of all subscriber and server data.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
is an Evolution of HLR(Home location register) which is in GSM.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
contain </span><b style="font-family: Verdana, sans-serif;">User Profiles</b><span style="font-family: Verdana, sans-serif;"> used by control layer</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
contain subscription information used by service layer</span></li>
</ul>
<br />
<div style="margin-bottom: 0cm; text-align: left;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div style="margin-bottom: 0cm; text-align: left;">
<span style="font-family: Verdana, sans-serif;"><b>User
profile</b> contains</span></div>
<div style="margin-bottom: 0cm; text-align: left;">
</div>
<ul style="text-align: left;">
<li><span style="font-family: Verdana, sans-serif;">.
User identity</span></li>
<li><span style="font-family: Verdana, sans-serif;">.allocated
s-cscf name</span></li>
<li><span style="font-family: Verdana, sans-serif;">.
Registration information and roaming profile</span></li>
<li><span style="font-family: Verdana, sans-serif;">.authentication
parameters</span></li>
<li><span style="font-family: Verdana, sans-serif;">.control
and service information</span></li>
</ul>
<br />
<div style="margin-bottom: 0cm; text-align: left;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div style="margin-bottom: 0cm; text-align: left;">
<span style="font-family: Verdana, sans-serif;"><b>SLF</b>
(<b>subscriber location function</b>):</span></div>
<div style="margin-bottom: 0cm; text-align: left;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div style="margin-bottom: 0cm; text-align: left;">
</div>
<ul style="text-align: left;">
<li><span style="font-family: Verdana, sans-serif;">An SLF is needed to map user address when multiple HSSs are
used.</span></li>
<li><span style="font-family: Verdana, sans-serif;">Network with Single HSS do not need SLF, On other hand networks with
more than one HSS require SLF.</span></li>
<li><span style="font-family: Verdana, sans-serif;">Both HSS and the SLF communicate through the Diameter
protocol</span></li>
</ul>
<br />
<div style="text-align: left;">
</div>
<div style="text-align: right;">
<span style="font-family: Verdana, sans-serif;"><br /></span></div>
<span style="font-family: Verdana, sans-serif;"><br />
</span><br />
<ol>
<li value="2"><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><b>CSCF(Call
session control function):</b>This is a sip server. There are
three types of CSCFs , depending on Functionalities they provide :-</span></div>
</li>
</ol>
<ol>
<li value="1"><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">PCSCF(proxy)
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">SCSCF(Serving)
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">ICSCF(interrogating)
</span></div>
</li>
</ol>
<div align="left" style="margin-bottom: 0cm; margin-left: 0.95cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm; margin-left: 0.95cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm; margin-left: 0.95cm;">
<b><span style="font-family: Verdana, sans-serif;">PCSCF:</span></b></div>
<ol>
<li value="1"><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">It
is the first point of contact between IMS terminal (UE) and
IMS network.
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"> Its
main functionalities are:-
</span></div>
</li>
</ol>
<ul type="disc">
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">It
establishes number of <b>IP
sec security associations </b>(
the ability to detect the content of message has changed since
its creation) towards the IMS terminal.
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">It
verifies the <b>correctness
of sip requests</b>
sent by IMS terminal and forwards sip messages to SCSCF.
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">it
forwards registration requests received from UE to
I-CSCF
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"> It
forwards requests and answer to the UE.
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">It
also Includes <b>Compressor
and de-compresso</b>r
of SIP messages.
</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">It
authenticate the User and asserts the identity of the
user to other nodes in the network.</span></div>
</li>
<li><div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
It
also include <b>PDF(policy
decision Function)</b>.It
is integrated with PCSCF or Standalone unit.PDF authorizes media
plane and manages Quality of service over media plane.
</span></div>
</li>
</ul>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"> 3)
The PCSCF may be located either in Visited network or Home network.</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">ICSCF:</span></b></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">I
CSCF is Sip proxy located at the edge of an
administrative domain.</span></b></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;">It's
IP Address is published in the DNS of the domain(using NAPTR and SRV
type of DNS records)</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
has an interface to SLF and HSS.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It Queries the HSS using Diameter cx Interface to retrieve the user
location.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It also implements interface to Application servers, to route
requests that are addressed</span><span style="font-family: Verdana, sans-serif;">to
services rather regular users.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It may optionally encrypt the parts of sip messages that
contain sensitive information about the domain, DNS names and
capacity. This functionality is called THIG(Topology hiding
inter-network gateway.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It is located in Home Network , In some special cases such as
ICSCF(THIG) it may be located in visited network as well.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">SCSCF:</span></b></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">S
CSCF is the central node of the signalling plane.</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;">It is a sip server always located in home network.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It Uses Diameter cx and DX to upload or download user profiles,
it has no local storage. All necessary information is stored in HSS.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
handles SIP registrations, Which allows to bind User location/IP
address and SIP address.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
sits on path of Signalling message and can inspect every message.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
decides to which application servers the sip message will be
forwarded, in order to provide services.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
provide Routing Services typically using ENUM lookups.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
enforces the policy of the network operator.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><b>MRF:(Media
Resource Function)</b> It provides a source of
media in the home network .</span></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;">It
is used for playing Announcements(audio/Video)</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
is used for Multimedia Conferencing( ex: Mixing audio streams)</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
is used for TTS(text-speech Conversion) and Speech recognition.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
is used for transcoding between different codec</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
is used for obtain statistics and do any sort of media analysis.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">It
is mainly divided into two types:</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">MRFC
and MRFP</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">MRFC(Media
resource function controller):</span></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;">It
is a signalling plane node that acts as a Sip user agent for
S-cscf and which controls the MRFP with a H.248 interface.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">MRFP:(Media
resource function processor):</span></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;">It
is media plane node that implements all media related
functions, such as playing and mixing media.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">MRF
is located in Home network
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">BGCF(Break
Out Gateway Control Function) :</span></b></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;">It
is a Sip server used for routing Calls between the IMS
terminal and PSTN phone.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
routes based on Telephone numbers.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It break out occurs in same network as the BGCF then the BGCF
select a MGCF that will be responsible for
internetworking with the PSTN and forwards the signalling
to MGCF. Other wise it forwards signalling to BGCF of another
operator.</span></li>
<li><span style="font-family: Verdana, sans-serif;">The
MGCF then receives the signalling from BGCF and manages the
internetworking with PSTN network.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">The
PSTN/CS Gateway:</span></b></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">The
internetworking with CS network is realized by several components for
signaling, media and control functions.</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><b>SGW(Signalling
Gateway): </b>
</span></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;">It
is an interface with signalling plane of CS network.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
performs Lower layer protocol conversion.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
transforms ISUP over MTP into ISUP over SCTP/IP.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">MGCF(Media
Gateway Control Function):</span></b></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;">It
performs call control protocol conversion between Sip and ISUP.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It interfaces SGW over SCTP.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It
controls MGW with a H.248(Megaco) interface.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">MGW:
(Media Gateway)</span></b></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;"><b>I</b>t is an interface with Media plane of CS network.</span></li>
<li><span style="font-family: Verdana, sans-serif;">It converts RTP to PCM</span></li>
<li><span style="font-family: Verdana, sans-serif;">It also performs media transcoding when Codecs doesn't match.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<b><span style="font-family: Verdana, sans-serif;">Application
Server(AS):</span></b></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">
AS is a sip entity that hosts and
executes services .</span></div>
<div align="left" style="margin-bottom: 0cm;">
</div>
<ul>
<li><span style="font-family: Verdana, sans-serif;">It
interface with the S-CSCF and I-CSCF using Sip and HSS using
Diameter.</span></li>
<li><span style="font-family: Verdana, sans-serif;">This allows third party providers and easy integration
and deployment of their value added services to the IMS
infrastructure.</span></li>
</ul>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"><br />
</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">There
are three different types of Application servers:-</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"> SIP
AS :- It hosts and executes IP multimedia services based on sip.</span></div>
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;"> OSA-SCS(open
service Access-service capability server):- It inherits OSA
capabilities to access the IMS securely from external network.</span></div>
<br />
<div align="left" style="margin-bottom: 0cm;">
<span style="font-family: Verdana, sans-serif;">IM
SSF(IP multimedia Service switching system Function):
It allows a GSM SCF(GSM service control function)to control an IMS
session. IMS SSF provides intelligent gateway functionality between
sip based IMS network an IN systems that use protocols such as
CAMEL,INAP,AIN and MAP.</span></div>
</div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com3tag:blogger.com,1999:blog-5206096933420493644.post-56361802240849952842015-04-29T00:39:00.003-07:002015-04-29T00:39:58.590-07:00VoIP Interview Questions and Answers -1<div dir="ltr" style="text-align: left;" trbidi="on">
<h1 class="western">
<strong><span style="font-size: small;">What
is VoIP?</span></strong></h1>
<span style="font-size: small;">VoIP stands for Voice Over
Internet Protocol or Voice Over IP. VoIP technology makes it possible
to convert analog voice signal into digital data and transmits it
over the Internet. (There are more likely possible pronunciations, as
well as vo-ipp, have been used, but generally, the single syllable -
voyp, as in voice - may be the most common within the industry.)</span><br />
<h1 class="western">
<a href="https://www.blogger.com/null" name="2"></a><strong><span style="font-size: small;">Why
VoIP is better than traditional phone services?</span></strong></h1>
<span style="font-size: small;">Due to its cost efficiency,
VoIP is more and more popular largely over traditional telepone
networks. VoIP cuts companies’ monthly phone bill by approximately
fifty percent. In addition to its cost efficiency, VoIP technology
ensures many advanced features, like conference calling, IVR, call
forwarding, automatic redial, call recording, etc. without extra
fees.</span><br />
<span style="font-size: small;">VoIP offers cheaper
international long distance rates that are generally one-tenth of
what is charged by traditional phone companies.</span><br />
<span style="font-size: small;">Due to its portability VoIP
is a really good option to avoid expensive hotel phone and cell phone
roaming charges. Only a high speed broadband connection (and a
plugged adapter) is needed and anyone can reach you at your local
number - independently of your location. Most of the times in-network
calls to other VoIP service subscribers are free even if the calling
parties are located in different parts of the world.</span><br />
<span style="font-size: small;">By using Internet
connection for both data traffic and voice calls, it is possible to
get rid of one monthly payment that are usually charged by most
Internet service providers. In addition, the Internet-based voice and
data transmission enables to avoid wireless roaming fees and long
distance rates.</span><br />
<h1 class="western">
<a href="https://www.blogger.com/null" name="3"></a><strong><span style="font-size: small;">What
are the advantages of VoIP?</span></strong></h1>
<span style="font-size: small;">In addition to its cost
efficiency, the feature-rich services and the metaphorical
disappearance of geographical boundaries as that were mentioned
above, VoIP has many other benefits as follows:</span><br />
<ul>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">VoIP
technology enables to detect and process touch tones and DTMF
responses </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">VoIP
systems can be automated easily </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">VoIP
systems allow to use more than one codec </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">VoIP
provides rich media service as more file formats can be used with
these systems </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">VoIP
ensures a much more flexible system than hardware based solutions </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Most
VoIP service providers provide a user control interface, typically a
web GUI, to their customers so that they can change features,
options, and services dynamically. </span>
</div>
</li>
<li><span style="font-size: small;">VoIP protocols run on
the application layer and are able to integrate or collaborate with
other applications such as email, web browser, instant messenger,
social-networking applications, etc. </span>
<br />
</li>
</ul>
<h1 class="western">
<a href="https://www.blogger.com/null" name="4"></a><strong><span style="font-size: small;">How
does VoIP work?</span></strong></h1>
<span style="font-size: small;">VoIP is usually based on
the SIP system that is the recognized standard. Any SIP compatible
device can talk to any other. Any SIP telephone can call another over
the Internet - any additional equipment is not needed for this. You
only need to plug your SIP phone into the Internet connection,
configure it then dial the other person. You can also connect
traditional analog phones to your VoIP network – in this case and
ATA device is needed.</span><br />
<span style="font-size: small;">In VoIP systems, your
analog voice is converted into packets of data (as little files), and
then transmitted to the recipient through the Internet and decoded
back into your voice at the other end. To make it quicker, these
packets are compressed before transmission, a bit like zipping a file
(it will be decompressed of course at the other party).</span><br />
<span style="font-size: small;">The advantages of
converting analog signals into digital data can be summarized as
follows: Digital format can be better controlled as it can be
compressed, routed, converted, etc. In addition, digital signals are
more noise tolerant than analog signals. Quality of Service (QoS)
ensures real-time errorless data streaming that allows interactive
data voice exchange as well.</span><br />
<h1 class="western">
<a href="https://www.blogger.com/null" name="5"></a><strong><span style="font-size: small;">What
is the actual cost of VoIP telephony?</span></strong></h1>
<span style="font-size: small;">If you only want to use
VoIP to communicate with other users in your VoIP network, you can do
that free of charge. If however you want to be able to use VoIP to
make and receive calls to/from people who are out of your VoIP
network or do not have VoIP, you will need to subscribe to a VoIP
service provider plan, and a gateway service may be also needed that
provides a bridge between VoIP and the conventional phone networks.</span><br />
<h1 class="western">
<a href="https://www.blogger.com/null" name="6"></a><strong><span style="font-size: small;">Is
it possible to replace the current traditional corporate PBX with a
VoIP one?</span></strong></h1>
<span style="font-size: small;">Definitely yes. VoIP is a
very cost-effective option for those companies who want to upgrade
their old PBX systems and VoIP ensures new features that traditional
PBX systems simply do not. To change to a VoIP system, companies can
buy an IP PBX, but it is also possible to add some VoIP
functionalities into an existing phone system.</span><br />
<h1 class="western">
<a href="https://www.blogger.com/null" name="7"></a><strong><span style="font-size: small;">What
kind of equipments do I need for creating a VoIP system?</span></strong></h1>
<span style="font-size: small;">Getting started with VoIP
is fairly simple. Assuming that you already have the 2 most important
ingredients (a Windows PC or Mac computer and a broadband Internet
connection), all you need to get started is the following:</span><br />
<ul>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Some
telephone or messaging software </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">A
microphone </span>
</div>
</li>
<li><span style="font-size: small;">Headphones or speakers
</span>
<br />
</li>
</ul>
<span style="font-size: small;">You can use a headset of
course rather than a microphone and speaker to leave your hands free.</span><br />
<span style="font-size: small;">In order to choose which
software to use, it is worth to consider the followings. Using voice
chat in G-Talk or Yahoo Messenger could be regarded as VoIP, so could
the highly publicised Skype; but these are all proprietary systems.
You can download them free of charge, but to talk to someone using
G-Talk, the person at the other end also needs G-Talk. The same
applies to Yahoo and, to a great extent, to Skype. They use their own
special system that is not open and will not connect to other systems
easily. So – especially for corporate using – it is rather
recommended to use such a softphone as <a href="http://www.counterpath.com/x-lite-download.html">ConterPath
X-Lite</a> with a SIP enabled IP PBX or access to an Internet Service
Provider.</span><br />
<h1 class="western">
<a href="https://www.blogger.com/null" name="8"></a><strong><span style="font-size: small;">Which
protocols describe VoIP connections?</span></strong></h1>
<span style="font-size: small;">VoIP has been implemented
in various ways using both proprietary protocols and
protocols based on open standards. You can see the VoIP
protocols below:</span><br />
<ul>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">H.323
</span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Media
Gateway Control Protocol (MGCP) </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Session
Initiation Protocol (SIP) </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">H.248 (also
known as Media Gateway Control (Megaco)) </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Real-time
Transport Protocol (RTP) </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Real-time
Transport Control Protocol (RTCP) </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Secure
Real-time Transport Protocol (SRTP) </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Session
Description Protocol (SDP) </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Inter-Asterisk
eXchange (IAX) </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Jingle XMPP VoIP
extensions </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<span style="font-size: small;">Skype
protocol </span>
</div>
</li>
<li><span style="font-size: small;">Teamspeak </span>
<br />
</li>
</ul>
<span style="font-size: small;">The most commonly used one
is SIP. Session Initiation Protocol is IETF signaling protocol
used for VOIP and other text and multimedia communication sessions
such as voice and video calls over Internet Protocol (IP).</span><br />
<span style="font-size: small;">SIP can be used for
creating, modifying and terminating two-party (unicast) or multiparty
(multicast) sessions. These sessions include Internet telephone
calls, multimedia distribution, multimedia conferences, instant
messaging, file transfer and online games.</span><br />
<strong><span style="font-size: 12pt;"><br /></span></strong>
<strong><span style="font-size: 12pt;">How
to get started on VoIP programming?</span></strong><br />
<span style="font-size: small;">The best way for creating
any VoIP application is using a VoIP development kit. These SDKs are
intended to provide a background support for your VoIP project by
offering prewritten VoIP components. It is quite effective and
comfortable to use these prewritten components, as you can save time
and money as well. (During the softphone development that will be
described below, Ozeki <a href="http://voip-sip-sdk.com/p_21-download-ozeki-voip-sip-sdk-voip.html">VoIP
SIP SDK</a> has been used for this purpose, that supports all the
.NET programming languages, so C# as well.)</span><br />
<span style="font-size: small;">For using these toolkits,
you have to add your preferred SDK as reference in you IDE. After you
have added it, you can reach all VoIP components that are needed to
be able to define the behaviour of such VoIP applications as
softphones, call recorders, IVR menu systems, software-based IP phone
systems (PBX), etc.</span><br />
<h1 class="western">
<a href="https://www.blogger.com/null" name="10"></a><strong><span style="font-size: small;">How
to make VoIP calls?</span></strong></h1>
<span style="font-size: small;">In order to be able to make
voice calls by using your own software application you need to
connect your system to the telephone network. This can be done in
three ways:</span><br />
<ul>
<li><div style="margin-bottom: 0cm;">
<strong><span style="font-size: small;">Option
1: Use a VoIP telephone adapter</span></strong><span style="font-size: small;"><br />A
VoIP telephone adapter is a hardware device that can be connected to
your Ethernet LAN or to your PC. There are VoIP telephone adapters
for GSM lines, for standard analog telephone lines and for ISDN
lines. When you connect this hardware to your Ethernet LAN, it will
receive an IP address. You need to configure this IP address in your
VOIP SDK. </span>
</div>
</li>
<li><div style="margin-bottom: 0cm;">
<strong><span style="font-size: small;">Option
#2: Use a SIP Account provided by VoIP telephone service
provider</span></strong><span style="font-size: small;"><br />There
are many VoIP telephone service providers worldwide that offer phone
service over the Internet. You need to subscribe to their service,
and you will receive a SIP account (including an IP address, a
username and a password). You need to configure the SIP account
details in your VoIP SDK. </span>
</div>
</li>
<li><strong><span style="font-size: small;">Option #3: Use
your existing office PBX if it is a VoIP system.</span></strong><span style="font-size: small;"><br />If
you already have an IP telephone system, you need to connect your
VoIP SDK to that over the LAN. The SDK can log in to the phone
system by using a SIP account and it can make telephone calls just
like any other </span>
<br />
</li>
</ul>
<br />
<div style="line-height: 100%; margin-bottom: 0cm;">
<br />
</div>
</div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com4tag:blogger.com,1999:blog-5206096933420493644.post-29789862566244748202015-04-29T00:21:00.001-07:002015-04-29T00:21:07.544-07:008x8 VoIP Test<div dir="ltr" style="text-align: left;" trbidi="on">
<span style="font-family: Verdana,sans-serif;">This Online Testing Utility will open a socket-connection to your
browser and pass simulated VoIP Traffic to your home/office computer.
This test measures the quality and performance of your Internet
connection between your home/office network and the 8x8 servers.</span><br />
<span style="font-family: Verdana,sans-serif;">click on below link:</span><br />
<br />
<span style="font-family: Verdana,sans-serif;"><a href="http://voiptest.8x8.com/">8x8 voip Test</a> </span></div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com0tag:blogger.com,1999:blog-5206096933420493644.post-16984081410643534492015-04-29T00:18:00.001-07:002015-04-29T00:19:10.366-07:00Internet and Voip Speed test<div dir="ltr" style="text-align: left;" trbidi="on">
<span style="font-family: Verdana,sans-serif;">This <a href="http://www.testskorosti.ru/" title="Speed Test">speed test</a> measures the quality and performance of Internet connections for Voice over IP by simulating
real VoIP sessions between our server and your computer. VoIP transmission consists of Session Initiation
Protocol (SIP) signaling and Real Time Protocol (RTP) udp data stream. We test only real-time part as
the most important factor of call quality. But first your Internet connection (download and upload) is
tested. After the test you can see results and comments depending on your connection measured parameters.
You may share your results in forums and webpages. You will see instructions how to do it when your test will be complete.
So let's select server near you and go to
<a href="http://www.speedtest.ph/" title="Test your Internet Speed using multi server tester">speed test</a> in your location.
You can also perform our <a alt="Test your ping, latency and speed" href="http://ping-test.net/">Ping test</a> and India <a href="http://speedtest.net.in/">Speed Test</a>.</span><br />
<span style="font-family: Verdana,sans-serif;"><br /></span>
<span style="font-family: Verdana,sans-serif;">If you have any problems with VoIP test, check if the JAVA SUN is installed in your web browser. <a href="http://www.java.com/download/index.jsp" rel="nofollow">Click here to check your JAVA installation</a>.
It is strongly recommended to be up-to-date with the newest JAVA
version. Sometimes your browser (Mozilla Firefox, Chorome) may ask to
permin VoIP TEST applet to run with JAVA - you ought to allow your
browser to do it. Otherwise, the "js/java problem" will appear in jitter
and packet loss result fields.
</span></div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com1tag:blogger.com,1999:blog-5206096933420493644.post-59401186183665674282014-09-26T00:16:00.002-07:002014-09-26T00:16:37.910-07:00How to install VM player on Ubuntu 14.04<div dir="ltr" style="text-align: left;" trbidi="on">
<div dir="ltr" id="docs-internal-guid-fce9ab89-b0d1-120f-6e5a-17654153a940" style="line-height: 1.15; margin-bottom: 0pt; margin-top: 0pt; text-align: center;">
<span style="background-color: transparent; color: black; font-family: Arial; font-size: 15px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">How to install VM player on Ubuntu 14.04</span></div>
<br /><br /><h1 dir="ltr" style="line-height: 1.1428571428571428; margin-bottom: 6pt; margin-top: 10pt;">
<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 28px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Introduction</span></h1>
<div dir="ltr" style="line-height: 1; margin-bottom: 6pt; margin-top: 0pt;">
<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">VMware Player allows you to run entire operating systems in a </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">virtual machine</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">, which runs on top of Ubuntu or Windows. To the </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">guest</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> operating system (the one running inside the virtual machine), it appears as though it were running on its own PC. The </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">host</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> operating system runs the VMware Player, which provides the </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">guest</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">with resources like network access. It can be downloaded for free from </span><a href="http://www.vmware.com/products/player/" style="text-decoration: none;"><span style="background-color: white; color: #dd4814; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">VMware</span></a><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">.</span></div>
<div dir="ltr" style="line-height: 1; margin-bottom: 6pt; margin-top: 0pt;">
<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Virtual machines configured with </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">an operating system and applications ready to perform a specific function</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> are called </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">virtual appliances</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">. An appliance can be created using certain VMware products, or you can download ready-made appliances. A wide variety of appliances (both certified and other-wise) are available from </span><a href="http://www.vmware.com/vmtn/appliances/" style="text-decoration: none;"><span style="background-color: white; color: #dd4814; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">VMware's Appliance Marketplace</span></a><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">.</span></div>
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<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">If you are a Windows (or other operating system) user looking for an official Ubuntu</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">appliance</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> to run, you will want to read only the last section.</span></div>
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<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">If you are an Ubuntu user who wishes to install or use the VMware Player </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">software</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">, continue reading.</span></div>
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<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 28px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Installing VMware Player on Ubuntu 14.04</span></h1>
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<li dir="ltr" style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; list-style-type: decimal; text-decoration: none; vertical-align: baseline;"><div dir="ltr" style="line-height: 1.5; margin-bottom: 12pt; margin-top: 0pt;">
<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Install required packages </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">build-essential</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> and </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">linux-headers</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">:</span></div>
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<li dir="ltr" style="background-color: white; color: #333333; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; list-style-type: lower-alpha; text-decoration: none; vertical-align: baseline;"><div dir="ltr" style="line-height: 1.5; margin-bottom: 12pt; margin-top: 0pt;">
<span style="background-color: #f3f3f3; color: #333333; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">sudo apt-get install build-essential linux-headers-$(uname -r)</span></div>
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</ol>
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<a href="https://my.vmware.com/web/vmware/free#desktop_end_user_computing/vmware_player/6_0" style="text-decoration: none;"><span style="background-color: white; color: #dd4814; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Download the latest VMware player</span></a><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> e.g. </span><span style="background-color: white; color: #333333; font-family: Verdana; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">VMware-Player-6.0.2-1744117.x86_64.bundle</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> (download the </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">bundle</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> version, not the </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: italic; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">rpm</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">one) and run it as root using gksudo. You'll get a graphical installer that installs VMware player for you.</span></div>
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</ol>
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<li dir="ltr" style="background-color: white; color: #333333; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; list-style-type: disc; margin-left: 32px; text-decoration: none; vertical-align: baseline;"><div dir="ltr" style="line-height: 1.2; margin-bottom: 12pt; margin-top: 0pt;">
<span style="background-color: #f3f3f3; color: #333333; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">gksudo bash ~/Downloads/VMware-Player-6.0.2-1744117.x86_64.bundle</span></div>
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</ul>
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<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">Note:</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> this assumes the location of your Downloads folder is </span><span style="background-color: white; color: #333333; font-family: Verdana; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">/home/$USER/Downloads</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">. *If nothing appears, you may need to make the file executable. You can do so with this command:</span></div>
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<span style="background-color: #f3f3f3; color: #333333; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">chmod +x ~/Downloads/VMware-Player-6.0.2-1744117.x86_64.bundle</span></div>
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<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">(again, with the assumption of your Downloads folder location). After completion, VMware player is installed and should show up in the menu under </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">Applications</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> → </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">System Tools</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> → </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">VMware Player</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> (for Unity users, it should come up in the search results for </span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">vmware player</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">).</span></div>
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<span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">As well, you may notice that when trying to create a new virtual machine, vmware player will complain on the terminal output(if it was started from the terminal as </span><span style="background-color: white; color: #333333; font-family: Verdana; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">vmplayer</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">) that:</span></div>
<div dir="ltr" style="line-height: 1.2; margin-bottom: 6pt; margin-top: 0pt;">
<span style="background-color: #f3f3f3; color: #333333; font-family: Verdana; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">VMware Player is installed, but it has not been (correctly) configured for your running kernel. To (re-)configure it, your system administrator must find and run "vmware-config.pl". For more information, please see the VMware Player documentation.</span></div>
<div dir="ltr" style="line-height: 1.5; margin-bottom: 6pt; margin-top: 0pt;">
<span style="background-color: white; color: #333333; font-family: Verdana; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">vmware-config.pl</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> is not present anymore in the latest vmware-player versions (seems to have been superseded by vmware-modconfig). If you have this problem you may instead need to check if you have a </span><span style="background-color: white; color: #333333; font-family: Verdana; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">/etc/vmware/not_configured</span><span style="background-color: white; color: #333333; font-family: Ubuntu; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> file and, if so, delete it.</span></div>
</div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com3tag:blogger.com,1999:blog-5206096933420493644.post-29289894886006378872014-09-23T01:52:00.002-07:002014-09-23T01:52:34.217-07:00Installing OpenIMSCore on Ubuntu<div dir="ltr" style="text-align: left;" trbidi="on">
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;">
</span></span><div>
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<div>
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;">First install the prerequisites for OpenIMSCore<br /><br /><span style="color: #38761d;">sudo apt-get install libcurl4-gnutls-dev<br />sudo apt-get install bison<br />sudo apt-get install curl<br />sudo
apt-get install debhelper cdbs lintian build-essential fakeroot
devscripts pbuilder dh-make debootstrap dpatch flex libxml2-dev
libmysqlclient15-dev ant docbook-to-man</span></span></span></div>
<div>
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;"><span style="color: #38761d;">sudo apt-get install ipsec-tools</span></span></span></div>
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<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;"><span style="color: #38761d;">sudo apt-get install subversion<br />sudo apt-get install mysql-server-5.5</span><br /><br />To install JDK 7:<br /><br /><span style="color: #38761d;">sudo add-apt-repository ppa:webupd8team/java<br />sudo apt-get update<br />sudo apt-get install oracle-java7-installer</span><br /><br />#It'll keep your java 7 installation up to date.<br /><br />#To automatically set up the Java 7 environment variables JAVA_HOME and PATH:<br /><br /><span style="color: #38761d;">sudo apt-get install oracle-java7-set-default</span><br /><br />------ Download openIMScore-------------------<br /><span style="color: #38761d;">sudo mkdir /opt/OpenIMSCore<br />cd /opt/OpenIMSCore<br /><br />sudo mkdir ser_ims<br />sudo svn checkout https://svn.code.sf.net/p/openimscore/code/ser_ims/trunk/</span> ser_ims<br /><br />------- problem on ubuntu 12 and solve it by this code----------------<br /><br /><span style="color: #38761d;">sudo sed -i '/include <curl\/types.h>/d' ser_ims/lib/lost/client.h</span><br /><br />------ installing IMScore-------<br /><span style="color: #38761d;">cd ser_ims<br />sudo make install-libs all<br />cd ..</span><br />------------- Compile FHoSS----------<br /><span style="color: #38761d;">sudo mkdir FHoSS<br />sudo svn checkout https://svn.code.sf.net/p/openimscore/code/FHoSS/trunk/ FHoSS<br /><br />cd FHoSS<br />sudo ant compile deploy<br />sudo sed -i 's/JAVA_HOME\/bin\/java/JAVA_HOME\/usr\/bin\/java/g' deploy/startup.sh<br />cd ..</span><br />--------------- Copy databases to sql server---------<br /><span style="color: #38761d;">mysql -u root -p -h localhost < ser_ims/cfg/icscf.sql<br />mysql -u root -p -h localhost < FHoSS/scripts/hss_db.sql<br />mysql -u root -p -h localhost < FHoSS/scripts/userdata.sql</span><br /><br />---- Configure DNS--------------</span></span></div>
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;">- To install DNS on ubuntu</span></span></div>
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;"><span style="color: #38761d;">sudo apt-get install bind9</span></span></span></div>
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;">-then<br /></span></span><div>
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;"><span style="color: #38761d;">sudo cp ser_ims/cfg/open-ims.dnszone /etc/bind/<br /><br />sudo sed -i '3azone "open-ims.test" {\n\ttype master;\n\tfile "\/etc\/bind\/open-ims.dnszone";\n};' /etc/bind/named.conf.local<br /><br />sudo
sed -i '2a127.0.0.1\topen-ims.test mobicents.open-ims.test
ue.open-ims.test presence.open-ims.test icscf.open-ims.test
scscf.open-ims.test pcscf.open-ims.test hss.open-ims.test' /etc/hosts</span><br /></span></span></div>
<div>
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;">- To restart DNS server</span></span></div>
<div>
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;"><span style="color: #38761d;">sudo /etc/init.d/bind9 restart</span></span></span></div>
<span style="font-size: small;"><span style="font-family: "Courier New",Courier,monospace;">------------- copy start up files-------------<br /><span style="color: #38761d;">sudo cp ser_ims/cfg/*.cfg ./<br />sudo cp ser_ims/cfg/*.xml ./<br />sudo cp ser_ims/cfg/*.sh ./</span><br /><br /><br />------ Start up your servers----------<br /><span style="color: #38761d;">sudo ./pcscf.sh<br />sudo ./scscf.sh<br />sudo ./icscf.sh<br />cd FHoSS/deploy<br />sudo sh startup.sh</span><br /><br />----- to access the FHoSS use this URL----------<br /><span style="color: #38761d;">http://hssAdmin:hss@localhost:8080/hss.web.console/</span></span></span></div>
TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com11tag:blogger.com,1999:blog-5206096933420493644.post-68118014700166196572014-09-23T00:15:00.001-07:002014-09-23T00:15:54.029-07:00How to Intsall SiPp on Ubuntu<div dir="ltr" style="text-align: left;" trbidi="on">
<div dir="ltr" id="docs-internal-guid-2558700f-a15d-9d5f-5b41-0afae92e2663" style="line-height: 1.15; margin-bottom: 0pt; margin-top: 0pt;">
<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">SIPp:</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">SIPp is a stress or performance test tool / traffic generator for the SIP protocol. It can work in both Scenarios (UAC /UAS) and establishes and releases multiple calls with the INVITE and BYE methods.It can also reads Custom XML scenario files describing from very simple to complex call flows. It features the dynamic display of statistics about running tests (call rate, round trip delay, and message statistics), periodic CSV statistics dumps, TCP and UDP over multiple sockets or multiplexed with retransmission management and dynamically adjustable call rates.</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">SIPp can also send media (RTP) traffic through RTP echo and RTP /pcap replay. Media can be audio or video.</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">SIPp can be used to test various real SIP equipment like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, SIP PBX, ... It is also very useful to emulate thousands of user agents calling your SIP system.</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">How to install:</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Using apt get: </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Sipp comes as ubuntu pakage, you can install sipp using following command.</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Update the package index:</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">sudo apt-get update</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Install sip-tester deb package:</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">sudo apt-get install sip-tester</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">When you run above command you can able run sipp as service from any on ubuntu. with above steps the sipp version installed will be </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> </span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">SIPp v3.2-PCAP, version unknown, built Dec 3 2011, 14:49:41</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">.</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">If you want latest SIPp version then you need to follow below steps.</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Install from Source code:</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Download the latest release from </span></div>
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<a href="https://github.com/SIPp/sipp/releases" style="text-decoration: none;"><span style="background-color: transparent; color: #1155cc; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: underline; vertical-align: baseline;">https://github.com/SIPp/sipp/releases</span></a></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Install Dependencies :</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Install following prerequisites before start compilation </span></div>
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<span style="background-color: #eeeeee; color: #222222; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">sudo apt-get install dh-autoreconf</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">sudo apt-get install ncurses-dev</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">sudo apt-get install build-essential</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">sudo apt-get install libssl-dev libpcap-dev</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">sudo apt-get install libncurses5-dev</span></div>
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<span style="background-color: #efefef; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">sudo apt-get install libsctp-dev lksctp-tools</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Download Source: Consider you have to install SIPp </span><a href="https://github.com/SIPp/sipp/tree/v3.4.1" style="text-decoration: none;"><span style="background-color: transparent; color: #1155cc; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: underline; vertical-align: baseline;">v3.4.1</span></a></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Install Dependencies.</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">cd /home/user/Desktop</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">mkdir sipp</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">Download the Source: wget </span><a href="https://github.com/SIPp/sipp/archive/v3.4.1.tar.gz" style="text-decoration: none;"><span style="background-color: transparent; color: #1155cc; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: underline; vertical-align: baseline;">https://github.com/SIPp/sipp/archive/v3.4.1.tar.gz</span></a></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">tar -xvzf v3.4.1.tar.gz</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">You have four options to compile SIPp: </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">Without TLS (Transport Layer Security), SCTP or PCAP support</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">: - This is the recommended setup if you don't need to handle SCTP, TLS or PCAP.</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># cd sipp-3.4.1</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># ./configure</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># make</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"><br class="kix-line-break" /></span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">With TLS support</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">, you must have installed</span><a href="http://www.openssl.org/" style="text-decoration: none;"><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> </span><span style="background-color: transparent; color: #1155cc; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: underline; vertical-align: baseline;">OpenSSL library</span></a><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> (>=0.9.8) (which may come with your system). Building SIPp consists only in adding the "--with-openssl" option to the configure command:</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># cd sipp-3.4.1</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># ./configure --with-openssl</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># make</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"><br class="kix-line-break" /></span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">With</span><a href="http://sipp.sourceforge.net/doc/reference.html#pcapplay" style="text-decoration: none;"><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;"> </span><span style="background-color: transparent; color: #1155cc; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: underline; vertical-align: baseline;">PCAP play</span></a><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;"> support</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">:</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># cd sipp-3.4.1</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># ./configure --with-pcap</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># make</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"><br class="kix-line-break" /></span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">With</span><a href="http://sipp.sourceforge.net/doc/reference.html#sctp" style="text-decoration: none;"><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;"> </span><span style="background-color: transparent; color: #1155cc; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: underline; vertical-align: baseline;">SCTP</span></a><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;"> support</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">:</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># cd sipp-3.4.1</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># ./configure --with-sctp</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># make</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"><br class="kix-line-break" /></span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">You can also combine these various options, e.g.:</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># cd sipp-3.4.1</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># ./configure --with-sctp --with-pcap --with-openssl</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"># make</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"><br class="kix-line-break" /></span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> After doing this you will get sipp binary in </span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">/home/user/Desktop/sipp/sipp-3.4.1</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> Directory.</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">to check Sipp version simply run </span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">./sipp -v</span><span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;"> in currecnt directory (sipp-3.4.1).</span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: normal; text-decoration: none; vertical-align: baseline;">If u have followed last option the version will be </span></div>
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<span style="background-color: transparent; color: black; font-family: 'Courier New'; font-size: 13px; font-style: normal; font-variant: normal; font-weight: bold; text-decoration: none; vertical-align: baseline;">SIPp v3.4.1-TLS-SCTP-PCAP-RTPSTREAM built Sep 23 2014, 12:39:45.</span></div>
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TechVickhttp://www.blogger.com/profile/02259622175123118322noreply@blogger.com9