a=sendrecy attribute present. (Note that older RFC 2543 compliant UAs may initiate hold using c=0.0.0.0.) For further examples of SDP use with SIP, see the SDP Offer Answer Examples document [8]. 7.2 RTP-Real-Time Transport Protocol Real-Time Transport Protocol [4] was developed to enable the transport of real-time packets containing voice, video, or other information over IP. RTP is defined by IETF Proposed Standard RFC 3550 (which updates RFC 1889). RTP does not provide any quality of service over the IP network-RTP pack- ets are handled the same as all other packets in an IP network. However, RTP allows for the detection of some of the impairments introduced by an IP net- work, such as: Packet loss; Variable transport delay; Out of sequence packet arrival; Asymmetric routing. As shown in the protocol stack of Figure 1.1, RTP is an application layer protocol that uses UDP for transport over IP. RTP is not text encoded, but uses a bit-oriented header similar to UDP and IP. RTP version 0 is only used by the vat audio tool for MBONE broadcasts. Version 1 was a pre-RFC imple- mentation and is not in use. The current RTP version 2 packet header is shown in Figure 7.1. RTP was designed to be very general; most of the headers are only loosely defined in the standard; the details are left to profile docu- ments. The 12 octets are defined as: SIP: Understanding the Session Initiation Protocol Sequence Number Timestamp SSRCI Figure 7.1 RTP packet header. Version (V): This 2-bit field is set to 2, the current version of RTP. Padding (P): If this bit is set, there are padding octets added to the end of the packet to make the packet a fixed length. This is most commonly used if the media stream is encrypted. Extension (X): If this bit is set, there is one additional extension follow- ing the header (giving a total header length of 14 octets). Extensions are defined by certain payload types. CSRC count (CC): This 4-bit field contains the number of content source identifiers (CSRC) are present following the header. This field