cess of setting up a call using into individual packets are most commonly used. SIP. As illustrated in Figure 3.6-8, the caller PC, which is The RTP payload formats are designed such that (i) an SIP user agent, initiates a call by sending an INVITE a payload format should be devised SO that the stream request to the called party. The message has to go being transported is still useful even in the presence of through the SIP server that serves the domain of the a moderate amount of packet loss and (ii) ideally, each called party. The SIP server is responsible for locating the packet should possibly be decoded and played out irre- addressee via a location service and routing the message spective of whether the preceding packets have been lost to the called party. Once the called party receives the or arrived late. INVITE request, it responses with 2 00 OK, which is sent Figure 3.6-7 shows examples of RTP packets gener- back to the caller. Then the caller sends an ACK directly ated for MPEG-4 video based on RFC 3016. Among to the called party, SO that the call is set up, and an RTP these examples, Figure 3.6-7(b) shows one of the most pipe is established for audio and video transmission. commonly used packetization methods that have the SIP, being a signaling protocol, is only responsible for best error-resilience capability. With this packetization initiating and establishing the session, but the actual method, one RTP packet contains one video packet. A communication is directly between the caller and the video packet contains resynchronization marker in- called party. formation at the beginning of the video payload. When the RTP packet containing the VOP header is lost, the other RTP packets can still be decoded due to the use of the Header Extension Code information in the video 3.6.3 Quality of service packet header. No extra RTP header field is necessary. For H.263 video, similar to MPEG-4 video described in 3.6.3.1 Minimizing latency Figure 3.6-7(b), RFC 2429 specifies that the PSC and slice header have to be