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

[TSAREGORODTSEV Yury (JIRA)]

TSAREGORODTSEV Yury created ASTERISK-26901:
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             Summary: 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: 14.3.0, 13.14.0
            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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