[asterisk-users] asterisk 13.18.0: pjsip: unnecessary 603 decline because of wrong codec decision
Antony.Stone at asterisk.open.source.it
Wed Nov 1 04:14:46 CDT 2017
On Wednesday 01 November 2017 at 08:10:36, Michael Maier wrote:
> 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
So, you're claiming to the provider that you can support all those codecs.
> - Provider sends 183 Session progress SDP: g729, alaw
> - Provider sends g729 rtp packages
> But: there is no license to transcode g729.
So, you shouldn't be offering it.
> 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.
Once the codec's been agreed on, between what the two sides offer to each
other, you can't change it later. Only offer what you're prepared to accept.
> My workaround is to disable g729 and things are working fine again for
> me (for this special case).
That's not a workaround - that's correct configuation.
If you don't have a G.729 licence, don't offer G.729 to the peer.
"It would appear we have reached the limits of what it is possible to achieve
with computer technology, although one should be careful with such statements;
they tend to sound pretty silly in five years."
- John von Neumann (1949)
Please reply to the list;
please *don't* CC me.
More information about the asterisk-users