1.1How are IP routers related to SIP servers? Are IP routers SIP aware?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.
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.
Yes, as long as the device that routes the SIP messages support IPV6.
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.
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...
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.
In theory no, but practically that's always the case.
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).
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.
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).
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.
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.