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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26901?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=236174#comment-236174 ] 

Asterisk Team commented on ASTERISK-26901:
------------------------------------------

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> 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
>    Affects Versions: 13.14.0, 14.3.0
>            Reporter: TSAREGORODTSEV Yury
>
> Testing opus between 2 asterisk hosts.
> On 1st host config with parameters, 2nd host use SDP values.
> 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
> 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
> REPLY:
> <------------>
> Audio is at 21428
> Adding codec opus to SDP
> <--- 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
> 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.



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