Case of ongoing audio call between UA1 and UA2 registered to different opensips
O1 and O2. Mediaproxy is started at O1 UA1. Call is going fine, UA1 sends a
re-invite for video, O2 receives request from O1 to relay to UA2 but chops off
a portion of the sdp containing the video port before sending it. Note the
m=video line.
This is trace the on O2
U 2015/09/24 03:26:25.751577 O1:5060 -> O2:5060
INVITE sip:[email protected]:5073 SIP/2.0.
Record-Route: <sip:19995678540@O1:5060;ftag=aBAabhRx-;lr>.
Via: SIP/2.0/UDP O1:5060;branch=z9hG4bK9293.6ddeddf3.0.
Via: SIP/2.0/UDP
172.16.0.18:5060;received=55.456.34.4;branch=z9hG4bK.xKjWy3C~v;rport=5060.
From: <sip:[email protected]>;tag=aBAabhRx-.
To: "csip" <sip:[email protected]>;tag=3m6-QAR.
CSeq: 22 INVITE.
Call-ID: 08syb7keM5.
Max-Forwards: 69.
Route: <sip:19995678540@O2:5060;ftag=aBAabhRx-;lr;did=da4.12eea932>.
Subject: Media change.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO, UPDATE.
Content-Type: application/sdp.
Content-Length: 397.
Contact:
<sip:[email protected]>;+sip.instance="<urn:uuid:457d1d6f-96a1-4e67-b2c0-9c1c50d6a5ab>".
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1).
Proxy-Authorization: Digest realm="mydomain.net",
nonce="56036d7900000001c89041633e412550979448da344e29d2",
username="19995675342", uri="sip:[email protected]:5073",
response="c6242475e3b3e6cc886bba1c93f0c73d".
.
v=0.
o=19995675342 3223 2488 IN IP4 55.456.34.4.
s=Talk.
c=IN IP4 199.999.999.812.
t=0 0.
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics.
m=audio 21238 RTP/AVP 96 0 101 97.
a=rtpmap:96 opus/48000/2.
a=fmtp:96 useinbandfec=1; stereo=0; sprop-stereo=0.
a=rtpmap:101 telephone-event/48000.
a=rtpmap:97 telephone-event/8000.
m=video 21242 RTP/AVP 96.
a=rtpmap:96 VP8/90000.
U 2015/09/24 03:26:25.752492 O2:5060 -> O1:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP O1:5060;branch=z9hG4bK9293.6ddeddf3.0.
Via: SIP/2.0/UDP
172.16.0.18:5060;received=55.456.34.4;branch=z9hG4bK.xKjWy3C~v;rport=5060.
From: <sip:[email protected]>;tag=aBAabhRx-.
To: "csip" <sip:[email protected]>;tag=3m6-QAR.
CSeq: 22 INVITE.
Call-ID: 08syb7keM5.
Server: OpenSIPS (2.1.1 (x86_64/linux)).
Content-Length: 0.
.
U 2015/09/24 03:26:25.752579 O2:5060 -> 55.456.34.4:5073
INVITE sip:[email protected]:5073 SIP/2.0.
Record-Route: <sip:19995678540@O2:5060;ftag=aBAabhRx-;lr>.
Record-Route: <sip:19995678540@O1:5060;ftag=aBAabhRx-;lr>.
Via: SIP/2.0/UDP O2:5060;branch=z9hG4bK9293.a7c918e.0.
Via: SIP/2.0/UDP O1:5060;branch=z9hG4bK9293.6ddeddf3.0.
Via: SIP/2.0/UDP
172.16.0.18:5060;received=55.456.34.4;branch=z9hG4bK.xKjWy3C~v;rport=5060.
From: <sip:[email protected]>;tag=aBAabhRx-.
To: "csip" <sip:[email protected]>;tag=3m6-QAR.
CSeq: 22 INVITE.
Call-ID: 08syb7keM5.
Max-Forwards: 68.
Subject: Media change.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO, UPDATE.
Content-Type: application/sdp.
Content-Length: 397.
Contact:
<sip:[email protected]>;+sip.instance="<urn:uuid:457d1d6f-96a1-4e67-b2c0-9c1c50d6a5ab>".
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1).
Proxy-Authorization: Digest realm="mydomain.net",
nonce="56036d7900000001c89041633e412550979448da344e29d2",
username="19995675342", uri="sip:[email protected]:5073",
response="c6242475e3b3e6cc886bba1c93f0c73d".
.
v=0.
o=19995675342 3223 2488 IN IP4 55.456.34.4.
s=Talk.
c=IN IP4 199.999.999.812.
t=0 0.
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics.
m=audio 21238 RTP/AVP 96 0 101 97.
a=rtpmap:96 opus/48000/2.
a=fmtp:96 useinbandfec=1; stereo=0; sprop-stereo=0.
a=rtpmap:101 telephone-event/48000.
a=rtpmap:97 telephone-event/8000.
m=vi
U 2015/09/24 03:26:25.966969 55.456.34.4:5073 -> O2:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP O2:5060;branch=z9hG4bK9293.a7c918e.0.
Via: SIP/2.0/UDP O1:5060;branch=z9hG4bK9293.6ddeddf3.0.
Via: SIP/2.0/UDP
172.16.0.18:5060;received=55.456.34.4;branch=z9hG4bK.xKjWy3C~v;rport=5060.
From: <sip:[email protected]>;tag=aBAabhRx-.
To: "csip" <sip:[email protected]>;tag=3m6-QAR.
Call-ID: 08syb7keM5.
CSeq: 22 INVITE.
.
---
Reply to this email directly or view it on GitHub:
https://github.com/OpenSIPS/opensips/issues/650
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