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

[Michael Maier]

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