[asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision

Michael Maier m1278468 at mailbox.org
Wed Nov 1 02:10:36 CDT 2017


I'm facing the following scenario:

- Initial call opened to asterisk: SDP g722,alaw,ulaw

- Outgoing call to provider started with Invite / SDP alaw, g726 and

- Provider sends 183 Session progress SDP: g729, alaw

- Provider sends g729 rtp packages

But: there is no license to transcode g729.

What is asterisk doing?
Asterisk decides to stop the call at all:
- Sends cancel to provider and 603 decline to internal caller.

What would have been correct?
It would have been correctly to switch to alaw as provider offered it, too.

My workaround is to disable g729 and things are working fine again for
me (for this special case).

I'm seeing here two basic problems:
1. Asterisk prevents a call which could have been working fine if
   asterisk would have done the switch to alaw which is offered by
2. Asterisk would have done completely unnecessary transcoding even if
   g729 transcoding would have been supported.

I would be glad, if Asterisk would take better care of what it's
deciding which codec to use. g729 was the codec with the lowermost
priority in the own offered SDP to the provider but it anyway accepts
this codec even though provider offers the own most desired alaw!


More information about the asterisk-users mailing list