[asterisk-users] codec negotiation or transcoding issue

[Lợi Đặng]

Asterisk might be unable to transcode rtp type from downstream to upstream,
or vice versa.
There's a bug reported here, for asterisk 12 or above, using chan_sip.
https://issues.asterisk.org/jira/browse/ASTERISK-25676
It says that you could avoid the bug by using chan_pjsip, but you still
encounter it?
Turn `core set debug 5` to see whether you have `Unsupported payload type
received` like I once did?
rgds,

On Wed, Mar 15, 2017 at 1:40 AM Faheem Muhammad <faheem2084 at gmail.com>
wrote:

> Hi,
> I'm facing strange issue while establishing inbound calls from SIP trunks.
> Provider A is sending (G729,Alaw,uLaw) offer and asterisk dial the peer
> with its preferred codec order(G729,aLaw, uLaw). The peer's phone send the
> codec list as (uLaw, speex) in 200 OK replay. The Peer's phone has selected
> only uLaw and speed in this case.
>
> Ideally Asterisk should establish the call on uLaw codec, but Asterisk
> establish the call with two codec for this call. For downstream RTP is
> established with G729 and for upstream RTP is established with uLaw codec.
> This behavior cause the one way audio for some phones like Eyebeam 1.5.9
> but Phonerlite latest version allow it and there is no audio issue.
>
> Is it normal SIP RFC 3261 behavior or there is something wrong with codec
> negotiation or transcoding?
>
> I'm using Asterisk 13.14.0 with realtime chan_pjsip compiled with bundled
> pjproject on centos 6.8_x64. I have tested it with Asterisk 11.x with
> chan_sip and it works fine.
>
> Please advise me how can I setup the call based on late negotiation
> mechanism?
>
> Thank you!
>
>
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