Triple-Play Service Deployment

Chapter 8: Troubleshooting VoIP
The fastest growing control plane protocol is Session Initiation
Protocol (SIP). SIP is published in RFC 2543. SIP messages can be
broken up into two major categories—messages from clients to
servers and messages from servers back to clients. Each message
has a message header that identifies the message type, calling
party, and called party.There are four basic message types: general
headers, entity headers, request headers, and response headers.
The other common signaling choice for residential based VoIP
services is Media Gateway Control Protocol (MGCP). MGCP is a
combination of SGCP (Simple Gateway Control Protocol) and IPDC
(IP Device Control Protocol). The main feature of the MGCP
protocol suite is the capability of breaking a telephony gateway
into two basic part—a call control and a media element.
VoIP codecs
Codecs are used to convert analog voice data to digital format for
transmission across a packet network and transform it back to
analog at the destination. Codecs can be implemented in
software, hardware, or both. Voice data generally uses one of two
types of codecs: waveform encoders and vocoders. Waveform
encoders attempt to accurately reproduce the input signal at the
expense of a higher bit rate. Vocoders represent the signal by
varying several parameters and use lower bit rates. As a rule, the
higher the bit rate used, the better the voice quality. Higher bit rate
codecs, on the other hand, generate more network traffic and
thereby reduce the overall call capacity of the network.
A number of different codec implementations are currently in use.
The only internationally agreed-upon standard—and the only one
required for compliance with the ITU-T’s H.323 standards—is
G.711 Pulse Code Modulation (PCM). Two variations of the
standard, called mu-law (for US use) and a-law (non-US use), are in
widespread use. All versions of G.711 signals operate at 64 Kbps.
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