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 www.chinaroby.com SIP The Session Initiation Protocol (SIP) has taken the world of VoIP by storm. Originally considered little more than an interesting idea, SIP now seems poised to dethrone the mighty H.323 as the VoIP protocol of choicecertainly at the endpoints of the network. The premise of SIP is that each end of a connection is a peer, and the protocol negotiates capabilities between them. What makes SIP compelling is that it is a relatively simple protocol, with a syntax similar to that of other familiar protocols such as HTTP and SMTP. SIP is supported in Asterisk with the chan_sip.so module. 8.2.2.1. History SIP was originally submitted to the Internet Engineering Task Force (IETF) in February of 1996 as "draft-ietf-mmusic-sip-00." The initial draft looked nothing like the SIP we know today and contained only a single request type: a call setup request. In March of 1999, after 11 revisions, SIP RFC 2543 was born. At first, SIP was all but ignored, as H.3

  The IAX protocol was developed by Digium for the purpose of communicating with other Asterisk servers (hence " the Inter-Asterisk eXchange protocol "). IAX is a transport protocol (much like SIP) that uses a single UDP port (4569) for both the channel signaling and Realtime Transport Protocol (RTP) streams. As discussed below, this makes it easier to firewall and more likely to work behind NAT. IAX also has the unique ability to trunk multiple sessions into one dataflow, which can be a tremendous bandwidth advantage when sending a lot of simultaneous channels to a remote box. Trunking allows multiple data streams to be represented with a single datagram header, to lower the overhead associated with individual channels. This helps to lower latency and reduce the processing power and bandwidth required, allowing the protocol to scale much more easily with a large number of active channels between endpoints. 8.2.1.2. Future Since IAX was optimized for voice, it has rece