Hi all.
Can i see any explanation of codec negotiation performed by Asterisk?
Is there any early-negotiation and late-negotiation concept in Asterisk?
Furthermore, i am looking for a solution in which i have to separate audio and video calls. What kind of settings and Client i should consider… and also starting video call within audio call. (i suppose this is relation to reINVITE and a new SDP session)
Thanks
This will depend on the technology. For SIP it uses early offer, but attempts to handle incoming late offers.
Note that SIP codec negotiation is not really negotiation (both sides say what they will accept), but Asterisk will not respond with a codec if it hasn’t been offered, and won’t use one in the RTP if it wasn’t in common between the two sides.
please refer this link:
aosabook.org/en/asterisk.html
We have some wrong codec offers when a client tries to start an audio call with user, say, 200.
Now , if 200 supports video codec as well, asterisk offers video in SDP while inviting 200.
This scenario is disturbing to switch a call from audio to video and video to audio.
What is the possible solution for this?