[Asterisk-Users] SIP Codecs

[Olle E. Johansson]

Ignace CARIA wrote:

> Is the SDP that negociate the codec to establish a voice communication?

Yes. The SDP, session description protocol, attachment to INVITE/ACK/OK
messages is the basis of selection of codecs. In Asterisk you either configure
codecs in the general section of SIP.conf or per user/peer.

In the [general] section, codecs are prioritized after what you prefer when
Asterisk sends an SIP INVITE to someone else.

In the [peer] section, there's no prioritization, you only list the codecs
Asterisk accepts from the offer from the other side. The other side made the
priority when sending the offer.

The SDP also includes other attributes regarding the media stream, like
port numbers for RTP, direction hints for the streams and RTP type used
for DTMF signalling in RTP.

/O