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VoIP: An In-Depth Analysis

Chapter Description

This chapter explains many of the issues facing Voice over IP (VoIP) and ways in which Cisco addresses these issues.

Transport Protocols

As explained in Chapter 6, "IP Tutorial," two main types of traffic ride upon Internet Protocol (IP): User Datagram Protocol (UDP) and Transmission Control Protocol (TCP). In general, you use TCP when you need a reliable connection and UDP when you need simplicity and reliability is not your chief concern.

Due to the time-sensitive nature of voice traffic, UDP/IP was the logical choice to carry voice. More information was needed on a packet-by-packet basis than UDP offered, however. So, for real-time or delay-sensitive traffic, the Internet Engineering Task Force (IETF) adopted RTP. VoIP rides on top of RTP, which rides on top of UDP. Therefore, VoIP is carried with an RTP/UDP/IP packet header.


RTP is the standard for transmitting delay-sensitive traffic across packet-based networks. RTP rides on top of UDP and IP. RTP gives receiving stations information that is not in the connectionless UDP/IP streams. As shown in Figure 7-9, two important bits of information are sequence number and timestamp. RTP uses the sequence information to determine whether the packets are arriving in order, and it uses the time-stamping information to determine the interarrival packet time (jitter).


Figure 7-9 Real-Time Transport Header

You can use RTP for media on demand, as well as for interactive services such as Internet telephony. RTP (refer to Figure 7-9) consists of a data part and a control part, the latter called RTP Control Protocol (RTCP).

The data part of RTP is a thin protocol that provides support for applications with real-time properties, such as continuous media (for example, audio and video), including timing reconstruction, loss detection, and content identification.

RTCP provides support for real-time conferencing of groups of any size within an Internet. This support includes source identification and support for gateways, such as audio and video bridges as well as multicast-to-unicast translators. It also offers QoS feedback from receivers to the multicast group, as well as support for the synchronization of different media streams.

Another new proposal defined in RFC 3611, RTP Control Protocol Extended Reports (RTCP XR), provides a rich set of data for VoIP management. The data for these extended reports can be provided by technology such as VQmon embedded into VoIP phones or gateways and sent periodically during the call to provide real-time feedback on voice quality. The reports generated present a very useful set of VoIP metrics data on network packet loss, RTP round trip delay, and so on.

Using RTP is important for real-time traffic, but a few drawbacks exist. The IP/RTP/UDP headers are 20, 8, and 12 bytes, respectively. This adds up to a 40-byte header, which is twice as big as the payload when using G.729 with two speech samples (20 ms). You can compress this large header to 2 or 4 bytes by using RTP Header Compression (CRTP). CRTP is covered in depth in Chapter 8.

Reliable User Data Protocol

Reliable User Data Protocol (RUDP) builds in some reliability to the connectionless UDP protocol. RUDP enables reliability without the need for a connection-based protocol such as TCP. The basic method of RUDP is to send multiples of the same packet and enable the receiving station to discard the unnecessary or redundant packets. This mechanism makes it more probable that one of the packets will make the journey from sender to receiver.

This also is known as forward error correction (FEC). Few implementations of FEC exist due to bandwidth considerations (a doubling or tripling of the amount of bandwidth used). Customers that have almost unlimited bandwidth, however, consider FEC a worthwhile mechanism to enhance reliability and voice quality.

Cisco currently utilizes RUDP in its PGW2200 product, which enables Signaling System 7 (SS7) to Q.931 over IP conversion. The Q.931 over IP is transmitted over RUDP.

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