[asterisk-bugs] [JIRA] (ASTERISK-26901) OPUS doesn't generate SDP accordingly to codecs.conf

[Ruben Iniguez (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26901?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=260959#comment-260959 ] 

Ruben Iniguez commented on ASTERISK-26901:
------------------------------------------

Ok. Sorry to have caused confusion. We are interested to have this configurability. Hope you can at some point figure out what is happening. If needed more symptoms form our side let me know. Thanks

> OPUS doesn't generate SDP accordingly to codecs.conf
> ----------------------------------------------------
>
>                 Key: ASTERISK-26901
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26901
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling, Codecs/codec_opus
>    Affects Versions: 13.14.0, 14.3.0
>            Reporter: TSAREGORODTSEV Yury
>            Severity: Major
>
> Testing opus between 2 asterisk hosts.
> On 1st host config with parameters, 2nd host use SDP values.
> {noformat}
> 1st host codecs.conf
> [opus]
> type=opus
> packet_loss=40
> complexity=10
> max_playback_rate=8000
> fec=1
> dtx=1
> 2nd host codecs.conf:
> [opus]
> type=opus
> packet_loss=40
> complexity=10
> {noformat}
> {noformat}
> Contact: <sip:test_x-lite2 at 10.0.151.1:5065>
> Call-ID: 3ae4b0d36d96054955ea4e440cd3bb30 at 10.0.151.1:5065
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 14.3.0
> Date: Tue, 28 Mar 2017 23:11:38 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 219
> v=0
> o=root 569990228 569990228 IN IP4 10.0.151.1
> s=Asterisk PBX 14.3.0
> c=IN IP4 10.0.151.1
> t=0 0
> m=audio 32446 RTP/AVP 107
> a=rtpmap:107 opus/48000/2
> a=fmtp:107 useinbandfec=1;usedtx=1
> a=maxptime:20
> a=sendrecv
> {noformat}
> REPLY:
> <------------>
> Audio is at 21428
> Adding codec opus to SDP
> {noformat}
> <--- Reliably Transmitting (no NAT) to 10.0.151.1:5065 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.151.1:5065;branch=z9hG4bK1ae768f6;received=10.0.151.1
> From: "test_x-lite2" <sip:test_x-lite2 at 10.0.151.1:5065>;tag=as2e7fddcc
> To: <sip:100 at xxxxxx:5065>;tag=as3579828e
> Call-ID: 259150181ae3c91f3a2521ab6d740cec at 10.0.151.1:5065
> CSeq: 102 INVITE
> Server: Asterisk PBX 14.3.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:100 at xxxxxxx>
> Content-Type: application/sdp
> Require: timer
> Content-Length: 212
> v=0
> o=root 914654045 914654045 IN IP4 xxxxx
> s=Asterisk PBX 14.3.0
> c=IN IP4 xxxxx
> t=0 0
> m=audio 21428 RTP/AVP 107
> a=rtpmap:107 opus/48000/2
> a=fmtp:107 usedtx=1
> a=maxptime:20
> a=sendrecv
> {noformat}
> as result max_playback_rate is missing in original SDP
> and useinbandfec in SDP answer. What a mystique ?
> Of course none of sides using playback rate 8000.
> packet_loss also seems doesn't do any effect.
> bandwidth always same no matter if its 0 or 100.



--
This message was sent by Atlassian JIRA
(v6.2#6252)