Asterisk 14.2.1 ARI codec negociation problem

Asterisk Asterisk Support
1

Hi guys, i’m using ari to create the channels then add to the bridge .

First i specify all my sip client with only one codec : opus.
No problem with add two channel to the bridge but when the third was added , it left immediately(see from the asterisk console log below) and then it led the caller channel left.

from the sip packet i can see when the client answered , the 200ok all sepcify to the opus codec(a=rtpmap:107 opus/48000/2)
from asterisk console i got a info:bridge swiched to softmix, i cant know whether this cause the problem.

Could someone help me out PLZZZZZZ!

Here are the logs when the third channel answered:
– Channel PJSIP/45398810014-00000030 joined ‘simple_bridge’ stasis-bridge <22>
>Bridge 22: switching from simple_bridge technology to softmix

[Nov 17 11:08:06] WARNING[32314]: translate.c:481 ast_translator_build_path: No translator path: (starting codec is not valid)
[Nov 17 11:08:06] WARNING[32314]: channel.c:5607 set_format: Unable to find a codec translation path: (slin) -> (h264|opus)
[Nov 17 11:08:06] WARNING[32314]: translate.c:481 ast_translator_build_path: No translator path: (starting codec is not valid)
[Nov 17 11:08:06] WARNING[32314]: channel.c:5607 set_format: Unable to find a codec translation path: (slin) -> (h264|opus)
[Nov 17 11:08:06] WARNING[32314]: translate.c:481 ast_translator_build_path: No translator path: (starting codec is not valid)
[Nov 17 11:08:06] WARNING[32314]: channel.c:5607 set_format: Unable to find a codec translation path: (slin) -> (h264|opus)
– Channel PJSIP/45398810014-00000030 left ‘softmix’ stasis-bridge <22>
> Bridge 22: switching from softmix technology to simple_bridge
– Channel PJSIP/45398810008-0000002e left ‘simple_bridge’ stasis-bridge <22>
– Channel PJSIP/45398810009-0000002f left ‘simple_bridge’ stasis-bridge <22>

Here r the endpoint info:
Endpoint: 45398810008/unknown In use 1 of inf
InAuth: 45398810008/45398810008
Aor: 45398810008 1
Channel: PJSIP/45398810008-00000027/Stasis Up 16:52:45
Exten: s CLCID: “45398810011:27:2” <>

ParameterName : ParameterValue

100rel : yes
accountcode :
acl :
aggregate_mwi : true
allow : (g722|ulaw|alaw|opus|h264)
allow_subscribe : true
allow_transfer : true
aors : 45398810008
asymmetric_rtp_codec : false
auth : 45398810008
bind_rtp_to_media_address : false
call_group :
callerid : unknown
callerid_privacy : allowed_not_screened
callerid_tag :
connected_line_method : invite
contact_acl :
context : default
cos_audio : 0
cos_video : 0
device_state_busy_at : 0
direct_media : false
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false

2

You appear not to have an opus codec, so it will work in pass through, but will fail the first time you try to conference. (I suppose you might get a similar effect if your codec options were unsupported by the opus codec, but I’m not certain that would happen so late.)

3

Ok, after some research , it seems clear but haven’t prove it.

  1. After reading the article ARI and Bridges found the section:
    Multi-party mixing - when there are more than two channels in a mixing bridge, Asterisk will **transcode** the media from each participant into signed linear, mix the media from all participants together into a new media frame, then write the media back out to all participants.
    So when i add the third channel into the bridge , it automatically does the transcode stuff.

  2. Knowing this, i searched why asterisk cant transcode the opus codec! Then i found this article asterisk audio Capabilities and seems that asterisk not handle opus codec by default.

  3. then i found this :[Asterisk14 OPUS transcoding](http://asterisk enable opus transcode)

Now i’m trying install codec_opus

4

Thanks & What a coincidence :smile: When i was editting my post i saw your reply.

5

Oh i forget another critical problem i meet caused by this scenario, after the third channel was kicked from the bridge

– Channel PJSIP/45398810014-00000030 left ‘softmix’ stasis-bridge <22>

the channel cant be destroyed!!! I tried with “channel request hangup” command in asterisk cli nothing happened!!!

then i use ARI DELETE /channels/{channelId}, nothing happened!!

Unless i restart my server!

It seems a bug of asterisk

issue1
another same question

6

The issue you referenced is not the same. Additionally you are using an older version of 14, and there were fixes in this area since then.

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