From SIPfoundry sipx, The Open Source SIP PBX for Linux - Calivia

[edit] How to set RTP codec preferences

sipXphone supports one DTMF (Dual Tone Multi-Frequency) codec and up to five audio codecs:

• telephone-event: Used to send out-of-band DTMF indications (as described in RFC 2833).
• pcmu: G.711 (u-law)
• pcma: G.711 (A-law)
• g729: G.729

Note: sipxPhone is ready to accommodate enhanced codecs from Global IP Sound such as the enhanced u-law and A-law versions of the G.711 codec. Because these codecs are not royalty free, they are only available in the supported version of sipXphone available from Pingtel. Note: This is the only difference between sipXphone from SIPfoundry and the supported version from Pingtel.

When sipXphone engages in call setup with another SIP user agent, they negotiate to determine which audio codec to use for the call. This often eliminates any need for codec transcoding, which increases audio quality.

PHONESET_RTP_CODECS

This parameter lists the codecs supported by sipXphone. The order in which audio codecs appear in this list indicates the preference to use when the phone negotiates with other SIP devices. Codecs that appear earlier in the list are preferred over those that appear later in the list. Defaults to:

PHONESET_RTP_CODECS : pcmu pcma g729 telephone-event

Tip: Since the telephone-event codec ID represents a DTMF codec, and not an audio codec, its use is not affected by its position in the list.

To change the value for this parameter you use the browser-based interface. In the browser-based interface, click Administration then Phone Configuration then scroll to the Additional Parameters text box at the bottom of this page. Enter PHONESET_RTP_CODECS, then each of the codec IDs separated by spaces. For example:

PHONESET_RTP_CODECS : g729 pcmu pcma telephone-event