Video is being activated systematically in audio call

Hi guys,

When video support is enabled in Asterisk (by setting videosupport=yes in sip.conf), and an audio call is made from a SIP client (using Teluu PJSIP Android library) to Asterisk server, the video is systematically being enabled on the callee side which is displaying an empty video view. Here’s the log from PJSIP app when making an audio call via Asterisk:

//Audio call

[CONFIRMED] t: “TestSIP” sip:TestSIP@XX.XX.XX.XX;tag=as76cb5bb7
Call time: 00h:00m:03s, 1st res in 147 ms, conn in 2528ms
#0 audio speex @8kHz, sendrecv, peer=-
SRTP status: Not active Crypto-suite:
ICE role: Unknown, state: Candidate Gathering, comp_cnt: 1
RX pt=110, last update:00h:00m:02.670s ago
total 143pkt 2.8KB (8.5KB +IP hdr) @avg=6.2Kbps/18.7Kbps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.375 17.965 25.375 21.750 5.500
TX pt=110, ptime=20, last update:never
total 80pkt 1.4KB (4.6KB +IP hdr) @avg=3.0Kbps/10.0Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000
#1 video H264, sendrecv, peer=-
SRTP status: Not active Crypto-suite:
ICE role: Unknown, state: Candidate Gathering, comp_cnt: 1
RX pt=99, size=352x288, fps=22.00, last update:00h:00m:03.633s ago
total 0pkt 0B (0B +IP hdr) @avg=0bps/0bps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
TX pt=99, size=112x112, fps=24.00, last update:never
total 83pkt 12.7KB (16.0KB +IP hdr) @avg=28.0Kbps/35.3Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000

We tried the same code to make an audio call via Linphone server. The callee did not display an empty video view. Here’s the log:

//Audio Call

[CONFIRMED] t: sip:testsip@sip.linphone.org;tag=acc79081-de20-4e76-b4bb-2f0cca9d12b7
Call time: 00h:00m:03s, 1st res in 103 ms, conn in 3067ms
#0 audio speex @16kHz, sendrecv, peer=XX.XX.XX.XX:54590
SRTP status: Not active Crypto-suite:
ICE role: Controlled, state: Negotiation Success, comp_cnt: 1
[0]: L:XX.XX.XX.XX:36872 § --> R:XX.XX.XX.XX:54590 (h)
[1]: L:0.0.0.0:0 (h) --> R:0.0.0.0:0 (h)
RX pt=98, last update:00h:00m:02.537s ago
total 135pkt 4.3KB (9.7KB +IP hdr) @avg=12.9Kbps/29.1Kbps
pkt loss=0 (0.0%), discrd=0 (0.0%), dup=0 (0.0%), reord=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.812 14.004 20.250 15.125 3.837
TX pt=98, ptime=20, last update:never
total 62pkt 1.8KB (4.3KB +IP hdr) @avg=5.6Kbps/13.1Kbps
pkt loss=0 (0.0%), dup=0 (0.0%), reorder=0 (0.0%)
(msec) min avg max last dev
loss period: 0.000 0.000 0.000 0.000 0.000
jitter : 0.000 0.000 0.000 0.000 0.000
RTT msec : 0.000 0.000 0.000 0.000 0.000
#1 video deactivated

We used the same clients on both sides with same config to call via Asterisk and Linphone. Linphone server has desactivated video as shown in the log, but Asterisk did not, causing the callee to show an empty video view although an explicit audio call was issued from the calling SIP client, using the following code:

MyCall call = new MyCall(account, -1);
CallOpParam prm = new CallOpParam(true);
prm.getOpt().setAudioCount(1);
prm.getOpt().setVideoCount((withVideo) ? 1 : 0);

withVideo having been false in both calls (to Linphone & Asterisk).

Is there anything we can set or do in Asterisk, to prevent enabling video on the callee side when a strict audio call is being made?

Thanks

Patrick

Expected behaviour. Asterisk has no feedback path for the codecs actually supported by the B side. In fact there may be more than one B side, but there can only be one 200 OK, so it would be wrong for it to refuse video just because the first B side doesn’t support it…

If you know that no B side in the call can handle video,you may be able to est channel variables to turn off the video codecs before you do anything than can answer the call.

Both sides can actually handle video but the call was explicitly defined as an audio call (as shown in the PJSIP source code provided earlier) when issued by the calling side. The issue is not happening with Linphone server, just Asterisk.

FYI, both sides support h264 only in the peer records in the realtime database and in the client software itself.

Are you saying that it is not possible to establish an audio-only call if the callee supports video?

Thanks

There are two calls. The codecs from the A side are used in setting up the B side, but only the fact that the B side has answered is fed back to A.

Dial says create a channel, passing it the selected codecs, but when B answers,only an AST_CONTROL_ANSWER frame is passed back, which only communicates that there has been an answer, so the call is accepted on the A side using all the codecs it had in common with its configuration in Asterisk.

For audio, this is not an issue, as Asterisk will transcode to resolve any incompatibilities.

Please bear with me as I am new to Asterisk. My question is that since A is telling B that it is requesting an audio-only call, why is B enabling video? And how can I prevent such a thing from happening? The A side, which initiated the call, is in audio mode only but the B side is responding by an unsollicited video support.

Thanks

I think I need the protocol logs, in particular, the SDP that Asterisk sends to the B side, in each case.

Here’s the log when peer A (peera) tried to make an audio call to peer B (peerb).

Thanks

<------------->
[2017-07-19 14:47:01] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:57559 --->
INVITE sip:peerb@xx.xx.xx.97:5060 SIP/2.0
v: SIP/2.0/UDP xx.xx.xx.130:57559;rport;branch=z9hG4bKPj8cbbe44a-b861-4e1c-bcb7-e2b88747f925
Max-Forwards: 70
f: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
t: sip:peerb@xx.xx.xx.97
m: <sip:peera@xx.xx.xx.130:57559;ob>;+sip.ice
i: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28240 INVITE
k: replaces, 100rel, timer, norefersub
x: 1800
Min-SE: 90
User-Agent: Pjsua2 Android 2.6
c: application/sdp
l: 528

v=0
o=- 3709457249 3709457249 IN IP4 xx.xx.xx.201
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 36437 RTP/AVP 98 97 99 104 18 3 0 8 9 96
c=IN IP4 xx.xx.xx.201
b=TIAS:64000
b=RS:0
b=RR:0
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ice-ufrag:4cbb6d80
a=ice-pwd:0b377c3e
a=candidate:Hac100bc9 1 UDP 2130706431 xx.xx.xx.201 36437 typ host
m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1
<------------->
[2017-07-19 14:47:01] VERBOSE[17524] chan_sip.c: --- (14 headers 24 lines) ---
[2017-07-19 14:47:01] VERBOSE[17524] chan_sip.c: Sending to xx.xx.xx.130:57559 (NAT)
[2017-07-19 14:47:01] VERBOSE[17524][C-00001259] chan_sip.c: Sending to xx.xx.xx.130:57559 (NAT)
[2017-07-19 14:47:01] VERBOSE[17524][C-00001259] chan_sip.c: Using INVITE request as basis request - bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
[2017-07-19 14:47:01] VERBOSE[17524][C-00001259] chan_sip.c: Found peer 'peera' for 'peera' from xx.xx.xx.130:57559
[2017-07-19 14:47:01] VERBOSE[17524][C-00001259] chan_sip.c: 
<--- Reliably Transmitting (NAT) to xx.xx.xx.130:57559 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xx.xx.xx.130:57559;branch=z9hG4bKPj8cbbe44a-b861-4e1c-bcb7-e2b88747f925;received=xx.xx.xx.130;rport=57559
From: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
To: sip:peerb@xx.xx.xx.97;tag=as607825e3
Call-ID: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28240 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05abaf3e"
Content-Length: 0


<------------>
[2017-07-19 14:47:01] VERBOSE[17524][C-00001259] chan_sip.c: Scheduling destruction of SIP dialog 'bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e' in 32000 ms (Method: INVITE)
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: Retransmitting #1 (NAT) to xx.xx.xx.130:57559:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xx.xx.xx.130:57559;branch=z9hG4bKPj8cbbe44a-b861-4e1c-bcb7-e2b88747f925;received=xx.xx.xx.130;rport=57559
From: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
To: sip:peerb@xx.xx.xx.97;tag=as607825e3
Call-ID: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28240 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05abaf3e"
Content-Length: 0


---
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:50215 --->

<------------->
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:57559 --->
INVITE sip:peerb@xx.xx.xx.97:5060 SIP/2.0
v: SIP/2.0/UDP xx.xx.xx.130:57559;rport;branch=z9hG4bKPj8cbbe44a-b861-4e1c-bcb7-e2b88747f925
Max-Forwards: 70
f: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
t: sip:peerb@xx.xx.xx.97
m: <sip:peera@xx.xx.xx.130:57559;ob>;+sip.ice
i: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28240 INVITE
k: replaces, 100rel, timer, norefersub
x: 1800
Min-SE: 90
User-Agent: Pjsua2 Android 2.6
c: application/sdp
l: 528

v=0
o=- 3709457249 3709457249 IN IP4 xx.xx.xx.201
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 36437 RTP/AVP 98 97 99 104 18 3 0 8 9 96
c=IN IP4 xx.xx.xx.201
b=TIAS:64000
b=RS:0
b=RR:0
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ice-ufrag:4cbb6d80
a=ice-pwd:0b377c3e
a=candidate:Hac100bc9 1 UDP 2130706431 xx.xx.xx.201 36437 typ host
m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1
<------------->
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: --- (14 headers 24 lines) ---
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Ignoring this INVITE request
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:57559 --->
ACK sip:peerb@xx.xx.xx.97:5060 SIP/2.0
v: SIP/2.0/UDP xx.xx.xx.130:57559;rport;branch=z9hG4bKPj8cbbe44a-b861-4e1c-bcb7-e2b88747f925
Max-Forwards: 70
f: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
t: sip:peerb@xx.xx.xx.97;tag=as607825e3
i: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28240 ACK
l: 0

<------------->
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: --- (8 headers 0 lines) ---
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:57559 --->
INVITE sip:peerb@xx.xx.xx.97:5060 SIP/2.0
v: SIP/2.0/UDP xx.xx.xx.130:57559;rport;branch=z9hG4bKPj97a4ae8d-5d8a-4d7b-bee7-25d2a6ac6238
Max-Forwards: 70
f: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
t: sip:peerb@xx.xx.xx.97
m: <sip:peera@xx.xx.xx.130:57559;ob>;+sip.ice
i: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28241 INVITE
k: replaces, 100rel, timer, norefersub
x: 1800
Min-SE: 90
User-Agent: Pjsua2 Android 2.6
Authorization: Digest username="peera", realm="asterisk", nonce="05abaf3e", uri="sip:peerb@xx.xx.xx.97:5060", response="5cf4baa9f4da813f4bb4cdf6d421fd39", algorithm=MD5
c: application/sdp
l: 528

v=0
o=- 3709457249 3709457249 IN IP4 xx.xx.xx.201
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 36437 RTP/AVP 98 97 99 104 18 3 0 8 9 96
c=IN IP4 xx.xx.xx.201
b=TIAS:64000
b=RS:0
b=RR:0
a=sendrecv
a=rtpmap:98 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:99 speex/32000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=ice-ufrag:4cbb6d80
a=ice-pwd:0b377c3e
a=candidate:Hac100bc9 1 UDP 2130706431 xx.xx.xx.201 36437 typ host
m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1
<------------->
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: --- (15 headers 24 lines) ---
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Sending to xx.xx.xx.130:57559 (NAT)
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Using INVITE request as basis request - bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found peer 'peera' for 'peera' from xx.xx.xx.130:57559
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 98
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 97
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 99
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 104
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 18
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 3
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 0
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 8
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 9
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 96
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found audio description format speex for ID 98
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found audio description format speex for ID 97
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found audio description format speex for ID 99
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found audio description format iLBC for ID 104
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Found audio description format telephone-event for ID 96
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Capabilities: us - (speex|ilbc|g722|g729|h264), peer - audio=(ulaw|gsm|alaw|g722|g729|speex|speex16|speex32|ilbc)/video=(nothing)/text=(nothing), combined - (speex|ilbc|g722|g729)
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Peer audio RTP is at port xx.xx.xx.201:36437
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Peer doesn't provide video
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: Looking for peerb in from-sip (domain xx.xx.xx.97)
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] sip/route.c: sip_route_dump: route/path hop: <sip:peera@xx.xx.xx.130:57559;ob>
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] chan_sip.c: 
<--- Transmitting (NAT) to xx.xx.xx.130:57559 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xx.xx.xx.130:57559;branch=z9hG4bKPj97a4ae8d-5d8a-4d7b-bee7-25d2a6ac6238;received=xx.xx.xx.130;rport=57559
From: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
To: sip:peerb@xx.xx.xx.97
Call-ID: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28241 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:peerb@xx.xx.xx.97:5060>
Content-Length: 0


<------------>
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: Audio is at 18656
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: Video is at xx.xx.xx.97:14878
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: Adding codec speex to SDP
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: Adding codec ilbc to SDP
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: Adding codec g722 to SDP
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: Adding codec g729 to SDP
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: Adding video codec h264 to SDP
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: Reliably Transmitting (NAT) to xx.xx.xx.130:33812:
INVITE sip:peerb@xx.xx.xx.130:33812;ob SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.97:5060;branch=z9hG4bK1fe71b0e;rport
Max-Forwards: 70
From: "peera" <sip:peera@xx.xx.xx.97>;tag=as229010bf
To: <sip:peerb@xx.xx.xx.130:33812;ob>
Contact: <sip:peera@xx.xx.xx.97:5060>
Call-ID: 599a50457d1d02cd662ca57108f50c71@xx.xx.xx.97:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Date: Wed, 19 Jul 2017 12:47:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 446

v=0
o=root 1263352823 1263352823 IN IP4 xx.xx.xx.97
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
c=IN IP4 xx.xx.xx.97
b=CT:384
t=0 0
m=audio 18656 RTP/AVP 110 97 9 18 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:30
a=sendrecv
m=video 14878 RTP/AVP 99
a=rtpmap:99 H264/90000
a=sendrecv

---
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:33812 --->
SIP/2.0 100 Trying
v: SIP/2.0/UDP xx.xx.xx.97:5060;rport=5060;received=xx.xx.xx.97;branch=z9hG4bK1fe71b0e
i: 599a50457d1d02cd662ca57108f50c71@xx.xx.xx.97:5060
f: "peera" <sip:peera@xx.xx.xx.97>;tag=as229010bf
t: <sip:peerb@xx.xx.xx.130;ob>
CSeq: 102 INVITE
l: 0

<------------->
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: --- (7 headers 0 lines) ---
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:33812 --->
SIP/2.0 180 Ringing
v: SIP/2.0/UDP xx.xx.xx.97:5060;rport=5060;received=xx.xx.xx.97;branch=z9hG4bK1fe71b0e
i: 599a50457d1d02cd662ca57108f50c71@xx.xx.xx.97:5060
f: "peera" <sip:peera@xx.xx.xx.97>;tag=as229010bf
t: <sip:peerb@xx.xx.xx.130;ob>;tag=fa22a9af-429e-4ac3-af13-5346f278998b
CSeq: 102 INVITE
m: <sip:peerb@xx.xx.xx.130:33812;ob>;+sip.ice
l: 0

<------------->
[2017-07-19 14:47:02] VERBOSE[17524] chan_sip.c: --- (8 headers 0 lines) ---
[2017-07-19 14:47:02] VERBOSE[17524][C-00001259] sip/route.c: sip_route_dump: route/path hop: <sip:peerb@xx.xx.xx.130:33812;ob>
[2017-07-19 14:47:02] VERBOSE[37854][C-00001259] chan_sip.c: 
<--- Transmitting (NAT) to xx.xx.xx.130:57559 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xx.xx.xx.130:57559;branch=z9hG4bKPj97a4ae8d-5d8a-4d7b-bee7-25d2a6ac6238;received=xx.xx.xx.130;rport=57559
From: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
To: sip:peerb@xx.xx.xx.97;tag=as1b2ddc3a
Call-ID: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28241 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:peerb@xx.xx.xx.97:5060>
Content-Length: 0


<------------>
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: Scheduling destruction of SIP dialog 'ZXUzU0VhYa' in 32000 ms (Method: REGISTER)
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:33812 --->

<------------->
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:33812 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP xx.xx.xx.97:5060;rport=5060;received=xx.xx.xx.97;branch=z9hG4bK1fe71b0e
i: 599a50457d1d02cd662ca57108f50c71@xx.xx.xx.97:5060
f: "peera" <sip:peera@xx.xx.xx.97>;tag=as229010bf
t: <sip:peerb@xx.xx.xx.130;ob>;tag=fa22a9af-429e-4ac3-af13-5346f278998b
CSeq: 102 INVITE
m: <sip:peerb@xx.xx.xx.130:33812;ob>;+sip.ice
k: replaces, 100rel, timer, norefersub
c: application/sdp
l: 405

v=0
o=- 3709457221 3709457222 IN IP4 xx.xx.14.30
s=pjmedia
b=AS:352
t=0 0
a=X-nat:0
m=audio 51934 RTP/AVP 110 101
c=IN IP4 xx.xx.14.30
b=TIAS:64000
a=sendrecv
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 46572 RTP/AVP 99
c=IN IP4 xx.xx.14.30
b=TIAS:256000
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42000a; packetization-mode=0
<------------->
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: --- (10 headers 19 lines) ---
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 110
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 101
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found audio description format speex for ID 110
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found audio description format telephone-event for ID 101
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP video format 99
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found video description format H264 for ID 99
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Capabilities: us - (speex|ilbc|g722|g729|h264), peer - audio=(speex)/video=(h264)/text=(nothing), combined - (speex|h264)
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Peer audio RTP is at port xx.xx.14.30:51934
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Peer video RTP is at port xx.xx.14.30:46572
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] sip/route.c: sip_route_dump: route/path hop: <sip:peerb@xx.xx.xx.130:33812;ob>
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Transmitting (NAT) to xx.xx.xx.130:33812:
ACK sip:peerb@xx.xx.xx.130:33812;ob SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.97:5060;branch=z9hG4bK667834f5;rport
Max-Forwards: 70
From: "peera" <sip:peera@xx.xx.xx.97>;tag=as229010bf
To: <sip:peerb@xx.xx.xx.130:33812;ob>;tag=fa22a9af-429e-4ac3-af13-5346f278998b
Contact: <sip:peera@xx.xx.xx.97:5060>
Call-ID: 599a50457d1d02cd662ca57108f50c71@xx.xx.xx.97:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Content-Length: 0


---
[2017-07-19 14:47:04] VERBOSE[37854][C-00001259] chan_sip.c: Audio is at 11598
[2017-07-19 14:47:04] VERBOSE[37854][C-00001259] chan_sip.c: Adding codec speex to SDP
[2017-07-19 14:47:04] VERBOSE[37854][C-00001259] chan_sip.c: Adding codec ilbc to SDP
[2017-07-19 14:47:04] VERBOSE[37854][C-00001259] chan_sip.c: Adding codec g722 to SDP
[2017-07-19 14:47:04] VERBOSE[37854][C-00001259] chan_sip.c: Adding codec g729 to SDP
[2017-07-19 14:47:04] VERBOSE[37854][C-00001259] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2017-07-19 14:47:04] VERBOSE[37854][C-00001259] chan_sip.c: 
<--- Reliably Transmitting (NAT) to xx.xx.xx.130:57559 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.130:57559;branch=z9hG4bKPj97a4ae8d-5d8a-4d7b-bee7-25d2a6ac6238;received=xx.xx.xx.130;rport=57559
From: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
To: sip:peerb@xx.xx.xx.97;tag=as1b2ddc3a
Call-ID: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28241 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:peerb@xx.xx.xx.97:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 392

v=0
o=root 592615378 592615378 IN IP4 xx.xx.xx.97
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
c=IN IP4 xx.xx.xx.97
t=0 0
m=audio 11598 RTP/AVP 97 104 9 18 96
a=rtpmap:97 speex/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:30
a=sendrecv
m=video 0 RTP/AVP 31

<------------>
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:42000 --->

<------------->
[2017-07-19 14:47:04] WARNING[37855][C-00001259] dsp.c: Inband DTMF is not supported on codec speex. Use RFC2833
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:57559 --->
ACK sip:peerb@xx.xx.xx.97:5060 SIP/2.0
v: SIP/2.0/UDP xx.xx.xx.130:57559;rport;branch=z9hG4bKPj6ff5cb71-d1fb-4fc8-af01-15c77ef2727e
Max-Forwards: 70
f: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
t: sip:peerb@xx.xx.xx.97;tag=as1b2ddc3a
i: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28241 ACK
l: 0

<------------->
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: --- (8 headers 0 lines) ---
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:57559 --->
INVITE sip:peerb@xx.xx.xx.97:5060 SIP/2.0
v: SIP/2.0/UDP xx.xx.xx.130:57559;rport;branch=z9hG4bKPj9d692304-76ad-4926-b5b2-3c17e005e26b
Max-Forwards: 70
f: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
t: sip:peerb@xx.xx.xx.97;tag=as1b2ddc3a
m: <sip:peera@xx.xx.xx.130:57559;ob>;+sip.ice
i: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28242 INVITE
k: replaces, 100rel, timer, norefersub
x: 1800;refresher=uas
Min-SE: 90
c: application/sdp
l: 298

v=0
o=- 3709457249 3709457250 IN IP4 xx.xx.xx.201
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 36437 RTP/AVP 97 96
c=IN IP4 xx.xx.xx.201
b=TIAS:64000
a=rtpmap:97 speex/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=sendrecv
m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1
a=inactive
<------------->
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: --- (13 headers 16 lines) ---
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Sending to xx.xx.xx.130:57559 (NAT)
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 97
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found RTP audio format 96
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found audio description format speex for ID 97
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Found audio description format telephone-event for ID 96
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Capabilities: us - (speex|ilbc|g722|g729|h264), peer - audio=(speex)/video=(nothing)/text=(nothing), combined - (speex)
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Peer audio RTP is at port xx.xx.xx.201:36437
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Peer doesn't provide video
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: 
<--- Transmitting (NAT) to xx.xx.xx.130:57559 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xx.xx.xx.130:57559;branch=z9hG4bKPj9d692304-76ad-4926-b5b2-3c17e005e26b;received=xx.xx.xx.130;rport=57559
From: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
To: sip:peerb@xx.xx.xx.97;tag=as1b2ddc3a
Call-ID: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28242 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:peerb@xx.xx.xx.97:5060>
Content-Length: 0


<------------>
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Audio is at 11598
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Adding codec speex to SDP
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Adding codec ilbc to SDP
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Adding codec g722 to SDP
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Adding codec g729 to SDP
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2017-07-19 14:47:04] VERBOSE[17524][C-00001259] chan_sip.c: 
<--- Reliably Transmitting (NAT) to xx.xx.xx.130:57559 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.130:57559;branch=z9hG4bKPj9d692304-76ad-4926-b5b2-3c17e005e26b;received=xx.xx.xx.130;rport=57559
From: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
To: sip:peerb@xx.xx.xx.97;tag=as1b2ddc3a
Call-ID: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28242 INVITE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:peerb@xx.xx.xx.97:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 392

v=0
o=root 592615378 592615379 IN IP4 xx.xx.xx.97
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
c=IN IP4 xx.xx.xx.97
t=0 0
m=audio 11598 RTP/AVP 97 104 9 18 96
a=rtpmap:97 speex/8000
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=0
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=maxptime:30
a=sendrecv
m=video 0 RTP/AVP 31

<------------>
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:55068 --->


<------------>
[2017-07-19 14:47:04] VERBOSE[17524] chan_sip.c: Scheduling destruction of SIP dialog 'f915cdb7-58c1-43e8-ac01-fbe30bcf0d3b' in 32000 ms (Method: REGISTER)
[2017-07-19 14:47:05] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:57559 --->
ACK sip:peerb@xx.xx.xx.97:5060 SIP/2.0
v: SIP/2.0/UDP xx.xx.xx.130:57559;rport;branch=z9hG4bKPja8412f69-a972-4d15-934e-1aed7738a4b2
Max-Forwards: 70
f: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
t: sip:peerb@xx.xx.xx.97;tag=as1b2ddc3a
i: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28242 ACK
l: 0

<------------->
[2017-07-19 14:47:05] VERBOSE[17524] chan_sip.c: --- (8 headers 0 lines) ---
[2017-07-19 14:47:05] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:57559 --->

<------------->
[2017-07-19 14:47:06] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:81.241.40.115:61689 --->

<------------->
[2017-07-19 14:47:07] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:38685 --->

<------------->
[2017-07-19 14:47:09] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:57559 --->
BYE sip:peerb@xx.xx.xx.97:5060 SIP/2.0
v: SIP/2.0/UDP xx.xx.xx.130:57559;rport;branch=z9hG4bKPj0b568a70-8787-4f76-adea-40499f4ef3b1
Max-Forwards: 70
f: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
t: sip:peerb@xx.xx.xx.97;tag=as1b2ddc3a
i: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28243 BYE
User-Agent: Pjsua2 Android 2.6
l: 0

<------------->
[2017-07-19 14:47:09] VERBOSE[17524] chan_sip.c: --- (9 headers 0 lines) ---
[2017-07-19 14:47:09] VERBOSE[17524][C-00001259] chan_sip.c: Sending to xx.xx.xx.130:57559 (NAT)
[2017-07-19 14:47:09] VERBOSE[17524][C-00001259] chan_sip.c: Scheduling destruction of SIP dialog 'bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e' in 32000 ms (Method: BYE)
[2017-07-19 14:47:09] VERBOSE[17524][C-00001259] chan_sip.c: 
<--- Transmitting (NAT) to xx.xx.xx.130:57559 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xx.xx.xx.130:57559;branch=z9hG4bKPj0b568a70-8787-4f76-adea-40499f4ef3b1;received=xx.xx.xx.130;rport=57559
From: sip:peera@xx.xx.xx.97;tag=9b6a7bb1-33ad-44e9-8bde-925fe26f0efe
To: sip:peerb@xx.xx.xx.97;tag=as1b2ddc3a
Call-ID: bd51e42b-adaa-44ac-a0cb-bb78bf3ed53e
CSeq: 28243 BYE
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[2017-07-19 14:47:09] VERBOSE[37855][C-00001259] chan_sip.c: Scheduling destruction of SIP dialog '599a50457d1d02cd662ca57108f50c71@xx.xx.xx.97:5060' in 32000 ms (Method: INVITE)
[2017-07-19 14:47:09] VERBOSE[37855][C-00001259] chan_sip.c: Reliably Transmitting (NAT) to xx.xx.xx.130:33812:
BYE sip:peerb@xx.xx.xx.130:33812;ob SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.97:5060;branch=z9hG4bK7f24aed5;rport
Max-Forwards: 70
From: "peera" <sip:peera@xx.xx.xx.97>;tag=as229010bf
To: <sip:peerb@xx.xx.xx.130:33812;ob>;tag=fa22a9af-429e-4ac3-af13-5346f278998b
Call-ID: 599a50457d1d02cd662ca57108f50c71@xx.xx.xx.97:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu4.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[2017-07-19 14:47:09] VERBOSE[17524] chan_sip.c: 
<--- SIP read from UDP:xx.xx.xx.130:33812 --->
SIP/2.0 200 OK
v: SIP/2.0/UDP xx.xx.xx.97:5060;rport=5060;received=xx.xx.xx.97;branch=z9hG4bK7f24aed5
i: 599a50457d1d02cd662ca57108f50c71@xx.xx.xx.97:5060
f: "peera" <sip:peera@xx.xx.xx.97>;tag=as229010bf
t: <sip:peerb@xx.xx.xx.130;ob>;tag=fa22a9af-429e-4ac3-af13-5346f278998b
CSeq: 103 BYE
l: 0

Please use </> when including logs!

Party A has not made an audio call;it has made a video call with the video initially on hold.

If it had made an audio call, these lines would not have been present:

m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1

If anything, the issue is that Asterisk doesn’t pass hold end to end. For audio, it would locally generate MoH.

1 Like

Sorry for the </>. Done.

And thanks for the info. I will investigate and see what I can do on the client side.

Hi David,

Something in the middle must be adding the m=video line. In fact, we made sure the outgoing message doesn’t include the line (by looking at the PJSIP client log after setting a bit flag to zero) but for some reason the line is still present in Asterisk’s log!

Any thoughts?

Patrick

What could possibly be modifying the message between the client and Asterisk?

Update: I used tPacketCapture to capture all packets during call. The outgoing packets contain no m=video lines. Still, the lines are appearing in the Asterisk log. This is really weird.

Somebody please help :slight_smile:

Can someone please confirm that sending an SDP with no m=video line to a peer supporting video via Asterisk is actually being logged without the m=video line? Could Asterisk itself be adding such a line to SDP? Could it be a bug?

Thanks