To provide the proper prioritization on a congested IP network, the IP network must have some knowledge of the applications.

Real-time Transport Protocol (RTP) is utilized in addition to a User Datagram Protocol (UDP)/IP header to provide timestamping . RTP runs atop UDP and IP and is commonly noted as RTP/UDP/IP. RTP is currently the cornerstone for carrying real-time traffic across IP networks. (Microsoft Netmeeting, for instance, utilizes RTP to carry audio and video communications.) To date, all VoIP signaling protocols utilize RTP/UDP/IP as their transport mechanism for voice traffic. Often, RTP packet flows are known as RTP streams . This nomenclature is used to describe the audio path.

In IP networks, it is common and normal for packet loss to occur. In fact, Transmission Control Protocol/Internet Protocol (TCP/IP) was built to utilize packet loss as a means of controlling the flow of packets. In TCP/IP, if a packet is lost, it is retransmitted. In most real-time applications, retransmission of a packet is worse than receiving a packet due to the time-sensitive nature of the information.

The ITU-T recommends a one-way delay of no more than 150 ms. In a Cisco VoIP network, the unidirectional delay might be 120 ms (currently, 65 ms to 85 ms of that 120-ms delay is derived from two Cisco VoIP gateways when using G.729). If the receiving station must request that a packet be re-transmitted, the delay will be too large, and large gaps and breaks in the conversation will occur.

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