[asterisk-users] pjsip: asterisk can't decide which codec to use

[Michael Maier]

On 05/12/2017 at 07:46 PM, Michael Maier wrote:

Forgot to mention: It's actual asterisk 13 branch from today (version 
before I tested, which has the same problem, was 13.15).


Regards,
Michael


> Hello!
>
> I'm facing completely choppy sound. The wireshark trace shows, that
> there are a lot of codec changes without any trigger (means no options
> or reinvite or any other package).
>
> Background:
> The call is initiated by asterisk and is received by the same asterisk
> conference room via
> Phone extension ->  asterisk -> provider A -> provider B -> asterisk.
>
> Asterisk initially sends invites using g722 and g711 and gets exactly
> this invite back as incoming call. The answer is g722,g711 in the ok sdp.
>
> Now, Asterisk can't decide, which codec to use. It frequently changes
> the codec just as it likes to apparently without any visible reason.
>
> [2017-05-11 17:28:03] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw
>
> [2017-05-11 17:28:03] DEBUG[5113][C-00000039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw
> [2017-05-11 17:28:04] DEBUG[5123][C-00000039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw
> [2017-05-11 17:28:04] DEBUG[5113][C-00000039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722
>
> [2017-05-11 17:28:04] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722
> [2017-05-11 17:28:04] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw
> [2017-05-11 17:28:13] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722
> [2017-05-11 17:28:13] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw
> [2017-05-11 17:28:19] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722
> [2017-05-11 17:28:19] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw
> [2017-05-11 17:28:23] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722
> [2017-05-11 17:28:23] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw
>
> [2017-05-11 17:28:23] DEBUG[5123][C-00000039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722
>
> [2017-05-11 17:28:23] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from alaw to g722
> [2017-05-11 17:28:28] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw
>
> [2017-05-11 17:28:28] DEBUG[5123][C-00000039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from g722 to alaw
>
> 0000003a -> inbound channel (callee)
> 00000039 -> outbound channel (caller)
>
>
> If I'm doing exactly the same call originated with another extension,
> there can't be seen these frequent changes. But the strange thing is,
> that in both cases the part between extension and asterisk doesn't show
> any codec changes ... .
>
> Deeper investigations show, that if the conference (callee) sends the
> first rtp package (-> g711 - should be g722), things are going choppy,
> if the extension (caller) sends the first package (g722), things are
> running stable.
>
>
> Any idea to convince asterisk always to use the first codec of ok sdp
> or how to convince asterisk to put only one codec to ok sdp (the first).
>
>
>
> Thanks,
> regards,
> Michael
>