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SIP Overview ---
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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.
SIP is a VoIP protocol – so does it carry voice packets?
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...
Do I always need to use a proxy server?
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.
Is the Microsoft messenger the only SIP based messenger tool?
Definitely not.
I want to build a SIP phone - is SIP all what it takes?
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.
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SIP Functionality ---
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Yes.
SIP supports, among others:
* call forwarding unconditional, busy, ...
* call transfer (call control spec)
* caller ID
* call hold
* 3-way conferences and multiparty conferencing (call control spec)
* call return ("*69")
* call park (with NOTIFY)
* follow-me
* find-me
* call waiting
* IVR systems
* multiple line presences
* call waiting
* camp on
* call queueing
* automatic call distribution
* do not disturb
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
recommendation
for business PBX-style services and other Internet phone
requirements.
Caller-ID
is provided by the From 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 signedby parameter, with the identity of the
company verified by a public key certificate similar to those used by
web sites.
Can a SIP-initiated session have zero or one participants?
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.
How do I charge/bill for Internet telephony using SIP?
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:
SIP services -
The
Authorization
header can be used to indicate a customer identity that associates a
SIP request with a billable entity.
Examples
of possibly chargeable SIP services include:
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Directory
services such as SIP proxy/redirect lookups;
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Customer
profile management; |
SIP
server operations can be charged based on server logs or, for
real-time billing, via AAA.
Media services - 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 RTSP.
Gateway services -
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.
Transport (network services) -
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.
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.
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 BYE
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 BYE
itself only suffices the end systems can be trusted by the network
provider not to keep sending packets.
Does
SIP carry DTMF? 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 (RFC 2833). 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.
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SIP Protocol Operation
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This is explained in Section 3 of RFC 2543. It refers to the section number in the HTTP/1.1 specification.
Do callers need to know the location of the location server?
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
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finger;
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LDAP/X.500;
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whois,
whois++;
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ph
and other local directories;
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shared
file systems with registration on login;
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local
SQL databases reached through TCP. |
Also,
callers don't register with the location server.
Which
parts of SIP are case-sensitive or case-insensitive?|
Method
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CS
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Header
field name
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CI
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Hide
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CI
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Accept-Encoding
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CI
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Accept
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CI
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Accept-Language
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CI
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Encoding
name (PCMU, L16, etc.)
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CI
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rfc1123-date
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CS
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A
call leg refers to the one-to-one signaling relationship between two
user agents (UAs). The Call-ID
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
INVITE;
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 Also
header.
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.
The
request URI is not used in call leg identification.
The
To
and From
field relate to local and remote in the following way. When Alice
sends a request on a call leg to Bob, the From
field contains the local address (Alice), and the To
field the remote address (Bob). When a request is received by Bob,
the To
field is matched to Bob's local address, and the From
field to the remote address (Alice).
The
CSeq
spaces in the two directions of a call leg are independent. Within a
single direction, the sequence number is incremented for each
transaction.
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.
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 Via header. (Igor Slepchin)
How can one recognize a retransmitted request?
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header
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retransmitted
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duplicate
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matching
response
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From
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same
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same
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same
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To
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same
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same
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same,
but tag may have been added
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Call-ID
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same
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same
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same
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request
URI
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same
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same
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same
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CSeq
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same
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same
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same
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Via
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-
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-
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must
be local host; check for branch parameter to identify which branch
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Looped
request are recognized by one or more of the following:
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The
server finds itself in the request's Via
list, including
any branch parameter.
(The server should compute the branch parameter so that it depends
on the request URI.)
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The
server is about to proxy the request to one of the hosts listed in
the Via
list. The same
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The
Max-Forward
count is decremented to zero.
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The
Expires
time has elapsed. |
The local registrar is either manually configured or discovered via DHCP (RFC 3361) . 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).
Is the domain of the request-URI and the To header always the same?
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 (To) might register under the Request-URI sip:atlanta.hiayh.org with the former as the To 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 REGISTER request is no longer forwarded once it has reached the server whose authoritative domain is the one listed in the Request-URI.
Not
per say. An ACK
is sent when a response retransmission is received. Reliability is
achieved because the response is retransmitted until an ACK
arrives, and the ACK
is retransmitted on response retransmissions. ACK
is only used for INVITE.
How
are BYE requests routed?
Since
a Contact
header MUST be present in INVITE
and 200, the BYE
will go directly to the user agent if there is no Record-Route
header. If there is a Record-Route,
it will traverse the list of proxies indicated there.
If
the caller decides to send a BYE
before receiving a 200 from the callee, the BYE
is being handled by the proxies just as the corresponding INVITE
was handled, i.e., it may be forked.
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 INVITE
and subsequent "re"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). Btw, In
the latter case ("re"INVITE),
the call remains, just any changes requests are cancelled.
What
is the relationship between the From, Contact, Via and
Record-Route/Route headers?All these headers determine how requests and responses are routed in a network of SIP proxy servers. Roughly, the distinction is:
From:
Used for subsequent requests if there is no Contact or Record-Route header. E.g., if Alice makes a call with From: Alice <alice@example.org> to Bob, an INVITE request from Bob to Alice would use alice@example.org as the To header and Request-URI.
Contact:
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 REGISTER requests and responses.
Record-Route/Route:
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.
Via:
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).
Generally, in short, requests should be sent to Route if present, Contact if there is no Route, From if there is no Contact.
How are URLs compared?
Two
SIP URLs are compared for equality according to the following rules:
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the
display name is ignored;
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tags
must match;
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user,
password, host, port and parameters of the URI must match. If a
component is omitted, it matches based on its default value.
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string
comparisons are case-insensitive;
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Characters
other than those in the "reserved" and "unsafe"
sets (see RFC 2396) are equivalent to their ""%"
HEX HEX" encoding.
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An
IP address that is the result of a DNS lookup on a hostname does
not match that hostname. |
Non-SIP URLs, such as tel:+12125551212 for a telephone number, may be used as request URIs in SIP INVITE 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 To 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 To start out the same.
Does SIP do admission control?
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.
Does SIP administer bandwidth?
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.
There is a proposal for an SDP extension (RFC 3312) 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).
Do I always need a proxy or redirect server?
No, two SIP endpoints can contact each other directly.
How does a caller find its proxy server?
Calls typically proceed directly to the callee's domain. For example, when calling alice@example.com the INVITE request would be sent to the SIP server for the domain example.com found via DNS.
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 REGISTER response or DHCP are under discussion.
What's the difference between a stateless and a stateful proxy server?
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 transaction state. Stateful implies transaction state, not call state.
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.
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 Record-Route header.
Proxies must be stateful if one of the following conditions hold:
- uses TCP,
- uses multicast,
- forks.
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.
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.
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.
How does a caller find the remote SIP client of the callee?
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 (server that supports the) desire transport, e.g. SIP over TCP)
The server located in this manner can then proxy or forward the call to another server.
How does SIP get through a firewall or NAT?
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 Socks V5. (Such server would normally be referred to as ALG (App. Layer Gateway))
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.)
As of 2006 the key solutions for NAT/firewall traversal are: SBC (Session Border Controller), ALG and using the STUN protocol (RFC 3489).
How does SIP do "call progress tones" or "ring back"?
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
100 Message received
100 Looking up number
100 Found number, looking up carrier according
to profile
100 Finding cheapest carrier which doesn't do
animal testing
100 Found carrier "AT&T"
100 Dialing number
180 Ringing
182 Queued, 3 people in front of you
182
Queued, 2 people in front of you
The
language of the status message should be determined based on the
Accept-Language
request header in the call.
A
183 (Session Progress) status response will appear in RFC2543bis. It
can be used for both progress tones as well as error messages.
One would use the 183 only
if you:|
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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
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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. |
However, this should only be done if the caller is likely a human being, as sending 183 would otherwise only delay failure handling.
Does SIP do keep-alive?
Originally it didn't, but now it does. See Session Timers (RFC 4028). (Thank you Vagelis Kalligeris for the typo correction!)
Why does SIP not have a Content-Transfer-Encoding header?
The Content-Transfer-Encoding 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 Content-Transfer-Encoding. SIP followed suit, and so does not use it either. Content-Encoding is used for things like compression, which is different. (J. Rosenberg)
See also RFC 2616 (HTTP/1.1), Section 19.4.5.
I want SIP to be more compact. What can I do?
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)
In general, more substantive savings are possible by using either Rohc/Signaling Compression (SigComp RFC 3320/3321/3486 etc) or IP payload compression (RFC 2393) 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.
What are the different addresses in SIP?
SIP INVITE requests involve three addresses:
- The host address where the request came from. Responses are sent back to the same host address, regardless of what the From header indicates. Note that different requests for the same call can come from different hosts.
- The From address contains the logical source of the request. It remains unmodified as a SIP request traverses proxies, for example. The From address may not be the same as the host address that generated the SIP request, although that's the typical case.
- 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 From
address.
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".
In what practical scenarios Call-Info header is(/can be) used?
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
in an INVITE, OPTIONS or REGISTER request. The header field parameter purpose indicates the purpose of the URI and may have the values
icon, info, card, or other IANA registered tokens. An example follows:
Call-Info: <http://www.code.com/my_picture.jpg>;purpose=icon. (Karimulla Sayed)
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Relationship To Other Protocols
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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 IETF and OMA standardization bodies.
What is the relationship between MGCP/Megaco/H.GCP and SIP?
The details of combining the two in a system are still being fleshed out. MGCP is a device control protocol, where a slave (gateway (MG)) is controlled by a master (media gateway controller (MGC), call agent). SIP may be used between 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.
What is SIP+ and how does it relate to SIP?
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.
How does SIP compare to H.323?
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.
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.
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.
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.
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)
Can SIP be used for Internet telephony gateways (ITGs)?
Yes, in two ways. First, it can indicate to the Internet-based caller that the callee is reachable via an ITG, via the Contact 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.
Can H.323 and SIP be used together?
Yes.
SIP can locate the called party and determine its capabilities,
including H.323. H.323 is then used to connect the two parties.
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.
How
do I interconnect Q.931 (ISDN signaling) and SIP?A gateway that initiates an ISDN call based on a SIP call or vice versa is reasonably straightforward.
How do I interconnect ISUP (SS7 signaling) and SIP?
Similar to the above. SIP-T and SIGTRAN provide standardization in this area.
What is sip-cgi and how does it relate to CPL?
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.
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.
In
its current version, CPL is only invoked when INVITE
requests and responses arrive, while sip-cgi can intercept any
request.
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.
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.
Note that CGI may be used as the execution environment for a CPL script. (Jonathan Rosenberg)
Is there a SIP interoperability certification? How can I test interoperability with others?
Check out the SIPit events (IOT) and the SIP Forum (certification).
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SIP Implementation/tools
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SIPp (this is not a typo...) (http://sipp.sourceforge.net) is a fantastic tool for accomplishing what I think you're after. [Rhys Ulerich]
Is there any tool that generates SIP log (call trace)?
Yes. check out Call Analyzers from: Ethereal (www.ethereal.com), Empirix (www.empirix.com), QuadTex (www.quadtexsys.com).
Are there any tools that would allow generation of graphical SIP call flows?
You might want to look at SIP Scenario: http://www.iptel.org/~sipsc/ 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.
Should
a terminal, which is intended to transmit calls to the PSTN, support
TRIP (Telephony Routing Over IP) or is it only the problem
of
the gateway?
TRIP is not needed in end
user phones or PC clients. It is used between servers facing each
other in a peering relationship between serviceproviders. Usage scenarios for TRIP are described fully in: http://www.ietf.org/rfc/rfc2871.txt
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:
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.
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.
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.
(-Jonathan R.)
