Only caller can watch the video

I am trying to make video call between two users

X-lite on mountain lion
Asterisk SVN-trunk-r373330

a)If UserA calls User B

UserA can watches the video but UserB not

b)If UserB calls User A
UserB can watches the video but UserA not.

There is the same message below on the console

[Oct 27 05:53:19] WARNING[6757][C-00000000]: chan_sip.c:9767 process_sdp: Ignoring video stream offer because port number is zero

SO I try ‘rtp set debug on’ and watch the console.

logs are like these below
(user A and UserB are in the same network and asterisk server is located out side of the network)

Got RTP packet from 219.110.194.:5068 (type 34, seq 005280, ts 38509780, len 000490)
Sent RTP packet to 219.110.194.
:5064 (type 34, seq 013929, ts 24417180, len 000490)
Sent RTP P2P packet to 219.110.194.:5062 (type 00, len 000160)
Sent RTP P2P packet to 219.110.194.
:5062 (type 00, len 000160)
Sent RTP P2P packet to 219.110.194.:5062 (type 00, len 000160)
Sent RTP P2P packet to 219.110.194.
:5062 (type 00, len 000160)
Got RTP packet from 219.110.194.:5068 (type 34, seq 005281, ts 38516980, len 000468)
Sent RTP packet to 219.110.194.
:5064 (type 34, seq 013930, ts 24424380, len 000468)
Sent RTP P2P packet to 219.110.194.:5062 (type 00, len 000160)
Sent RTP P2P packet to 219.110.194.
:5062 (type 00, len 000160)
.
.
.

Please help me…

RTP debugging is irrelevant. The video is being rejected, by the one of the parties, at the SIP/SDP level.

The message means that the peer understands the requested video codec, but is rejecting its use. The SDP exchange may give some clues.

Hi David55

Thanks,
I tried
’rtp set debug on’ and called from ‘teacher’ to ‘student’ and checked the SDP Exchange information

my log is here.
ow.ly/eODrX

It is very long but, there are messages like these.

[Oct 28 00:09:09] WARNING[9666][C-00000002]: chan_sip.c:9767 process_sdp: Ignoring video stream offer because port number is zero

Capabilities: us - (ulaw|alaw|h261|h263|h263p|h264), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

Peer audio RTP is at port 192.168.0.11:5062

Peer doesn't provide video

I think it must be hint.
but in my user.conf teacher and student setting are definitely the same.

I can’t find out why it says

peer - audio=(ulaw|alaw)/video=(nothing)

Thanks very much ,please give me some advice.

in user.conf

[student]
nat=yes
type=peer
defaultuser=student
context=default
secret=student
canreinvite=no
host=dynamic
dtmfmode=rfc2833
callgroup=1
pickupgroup=1
mailbox=203
videosupport=yes
disallow=all
allow=ulaw
allow=alaw
allow=h261
allow=h263
allow=h263p
allow=h264


[teacher]
nat=yes
type=peer
defaultuser=teacher
context=default
secret=teacher
canreinvite=no
host=dynamic
dtmfmode=rfc2833
callgroup=1
pickupgroup=1
mailbox=203
videosupport=yes
disallow=all
allow=ulaw
allow=alaw
allow=h261
allow=h263
allow=h263p
allow=h264

It is not giving any video, because there are no video codecs in the media line being processed; it is one for audio only. It looks like the line for video indicates a rejection.

Please look at en.wikipedia.org/wiki/Session_De … n_Protocol and find all the lines of the form described, for both parties.

HI David55
Thanks for your kindly help.

I can guess this issue is because of codec. as you told me.

in this logs.
ow.ly/eODrX

a)Teacher is dialing to student

    -- Executing [student@default:1] Dial("SIP/teacher-00000004", "SIP/student") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 13738
Video is at 49.212.184.**:11004
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding video codec 200001 (h261) to SDP
Adding video codec 200002 (h263) to SDP
Adding video codec 200003 (h263p) to SDP Adding video codec 200004 (h264) to SDP

b)Student answering the call

-- SIP/student-00000005 answered SIP/teacher-00000004 Audio is at 18838 Adding codec 100003 (ulaw) to SDP Adding codec 100004 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP

There are no ‘Adding video codec messages’ in b)

and there are log like these below

v=0 o=- 1351350549515669 1 IN IP4 192.168.0.11 s=CounterPath X-Lite 5.0.0 c=IN IP4 192.168.0.11 b=AS:1638 t=0 0 m=audio 5062 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 0 RTP/AVP 31 34 98 99

m=video 0 RTP/AVP 31 34 98 99
it says port0 for video.

But I don’t know why this happens.

When I exchange Teacher and Student
(Student calls teacher)
Student can watch videos.

so it doesn’t depend on the each user setting in user.conf.

Is there any place to configure?

It would help to have the SDP from both sides. If this is the second one, the Counterpath is rejecting video, but one needs to see the first one to see what video options were offered.

There are multiple messages so it is difficult which message is the issue point.
I summarized SDP messages from logs.
Please look and help me?

There are 9 messages and only message (d) says port 0

teacher calles and student answers.
teacher watch the video,but student cannot

192.168.0.10 is teacher
192.168.0.11 is student.
49.212.184.** is asterisk server

a)message a
from teacher(server) to student(server) contacted by teacher(local)

Contact: <sip:teacher@192.168.0.10:5060>
To: <sip:student@49.212.184.**>
From: "teacher"<sip:teacher@49.212.184.**>;tag=7b3b9e76

v=0
o=- 1351350542767463 1 IN IP4 192.168.0.10
s=CounterPath X-Lite 5.0.0
c=IN IP4 192.168.0.10
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15 
a=sendrecv

b)message b

from teacher(server) to student(server) contacted by teacher(local)

Contact: <sip:teacher@192.168.0.10:5060>
To: <sip:student@49.212.184.**>
From: "teacher"<sip:teacher@49.212.184.**>;tag=7b3b9e76
v=0
o=- 1351350542767463 1 IN IP4 192.168.0.10
s=CounterPath X-Lite 5.0.0
c=IN IP4 192.168.0.10
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv 

c)message c

from teacher(server) to student(local) contacted by teacher(server)

From: "New User" <sip:teacher@49.212.184.**>;tag=as6712df98
To: <sip:student@192.168.0.11:5060;rinstance=c5113327fb520b4d>
Contact: <sip:teacher@49.212.184.**:5060>

v=0
o=root 1222860354 1222860354 IN IP4 49.212.184.**
s=Asterisk PBX SVN-trunk-r373330
c=IN IP4 49.212.184.**
b=CT:384
t=0 0
m=audio 13738 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:2361e8514a43cba5038bb7e4765a1b2a
a=ice-pwd:25144ca13e68cc60557cafc76fde0776
a=candidate:H31d4b83a 1 UDP 2130706431 49.212.184.** 13738 typ host
a=candidate:H31d4b83a 2 UDP 2130706430 49.212.184.** 13739 typ host
a=sendrecv
m=video 11004 RTP/AVP 31 34 98 99
a=ice-ufrag:6f1c89135c54c11f737f133616711c7a
a=ice-pwd:51cb527743bb0f520f42cffa254d3760
a=candidate:H31d4b83a 1 UDP 2130706431 49.212.184.** 11004 typ host
a=candidate:H31d4b83a 2 UDP 2130706430 49.212.184.** 11005 typ host
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=fmtp:34 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0 
a=sendrecv

d)message d

from teacher(server) to student(local) contacted by student(local)

Contact: <sip:student@192.168.0.11:5060;rinstance=c5113327fb520b4d>
To: <sip:student@192.168.0.11:5060;rinstance=c5113327fb520b4d>;tag=62c7ca16
From: "New User"<sip:teacher@49.212.184.**>;tag=as6712df98

v=0
o=- 1351350549515669 1 IN IP4 192.168.0.11
s=CounterPath X-Lite 5.0.0
c=IN IP4 192.168.0.11
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv 
m=video 0 RTP/AVP 31 34 98 99

e)message e

from teacher(server) to student(server)contacted by student(server)

From: "teacher"<sip:teacher@49.212.184.**>;tag=7b3b9e76
To: <sip:student@49.212.184.**>;tag=as52d6686b
Contact: <sip:student@49.212.184.**:5060>

v=0
o=root 1943878915 1943878915 IN IP4 49.212.184.**
s=Asterisk PBX SVN-trunk-r373330
c=IN IP4 49.212.184.**
t=0 0
m=audio 18838 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:7a8521b133dd00ab58bed4b226473d40
a=ice-pwd:5b10bfef4be12d945a2447660c6751d0
a=candidate:H31d4b83a 1 UDP 2130706431 49.212.184.** 18838 typ host
a=candidate:H31d4b83a 2 UDP 2130706430 49.212.184.** 18839 typ host 
a=sendrecv

f) message f
from teacher(server) to student(server) contacted by teacher(local)

Contact: <sip:teacher@192.168.0.10:5060>
To: <sip:student@49.212.184.**>;tag=as52d6686b
From: "teacher"<sip:teacher@49.212.184.**>;tag=7b3b9e76

v=0
o=- 1351350542767463 2 IN IP4 192.168.0.10
s=CounterPath X-Lite 5.0.0
c=IN IP4 192.168.0.10
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 5064 RTP/AVP 34 115
b=TIAS:1574400
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;VGA=1;CIF4=1
a=rtpmap:115 H263-1998/90000
a=fmtp:115 QCIF=1;CUSTOM=160,120,1;CIF=1;CUSTOM=320,240,1;VGA=1;CIF4=1;I=1;J=1;T=1 
a=sendrecv

g)message g

from teacher(server)to student(server) contacted by student(server)

From: "teacher"<sip:teacher@49.212.184.**>;tag=7b3b9e76
To: <sip:student@49.212.184.**>;tag=as52d6686b
Contact: <sip:student@49.212.184.**:5060>

v=0
o=root 1943878915 1943878916 IN IP4 49.212.184.**
s=Asterisk PBX SVN-trunk-r373330
c=IN IP4 49.212.184.**
b=CT:384
t=0 0
m=audio 18838 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:7a8521b133dd00ab58bed4b226473d40
a=ice-pwd:5b10bfef4be12d945a2447660c6751d0
a=candidate:H31d4b83a 1 UDP 2130706431 49.212.184.** 18838 typ host
a=candidate:H31d4b83a 2 UDP 2130706430 49.212.184.** 18839 typ host
a=sendrecv
m=video 11500 RTP/AVP 34 115
a=ice-ufrag:0c445c8838923a2c323078db55c222c7
a=ice-pwd:69917c9d2c36d9bc1e967c505c0980c1
a=candidate:H31d4b83a 1 UDP 2130706431 49.212.184.** 11500 typ host
a=candidate:H31d4b83a 2 UDP 2130706430 49.212.184.** 11501 typ host
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;CIF4=1;VGA=1;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:115 h263-1998/90000
a=fmtp:115 QCIF=1;CIF=1;CIF4=1;VGA=1;CUSTOM=320,240,1;F=0;I=1;J=1;T=1;K=0;N=0;BPP=0;HRD=0
a=sendrecv 

h)message h

from student(local) to teacher(server) contacted by student(local)

Contact: <sip:student@192.168.0.11:5060;rinstance=c5113327fb520b4d>
To: "New User"<sip:teacher@49.212.184.**>;tag=as6712df98
From: <sip:student@192.168.0.11:5060;rinstance=c5113327fb520b4d>;tag=62c7ca16

v=0
o=- 1351350549515669 2 IN IP4 192.168.0.11
s=CounterPath X-Lite 5.0.0
c=IN IP4 192.168.0.11
b=AS:1638
t=0 0
m=audio 5062 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
m=video 5064 RTP/AVP 31 34 98 99 34 115
b=TIAS:1574400
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;VGA=1;CIF4=1
a=rtpmap:115 H263-1998/90000
a=fmtp:115 QCIF=1;CUSTOM=160,120,1;CIF=1;CUSTOM=320,240,1;VGA=1;CIF4=1;I=1;J=1;T=1 
a=sendrecv

i)message i

from stundet(local) to teacher(server) contacted by teacher(server)

From: <sip:student@192.168.0.11:5060;rinstance=c5113327fb520b4d>;tag=62c7ca16
To: "New User"<sip:teacher@49.212.184.**>;tag=as6712df98
Contact: <sip:teacher@49.212.184.**:5060>
Call-ID: 3b30eeff67cc0f8110afff3c074036a1@49.212.184.**:5060

v=0
o=root 1222860354 1222860355 IN IP4 49.212.184.**
s=Asterisk PBX SVN-trunk-r373330
c=IN IP4 49.212.184.**
b=CT:384
t=0 0
m=audio 13738 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:2361e8514a43cba5038bb7e4765a1b2a
a=ice-pwd:25144ca13e68cc60557cafc76fde0776
a=candidate:H31d4b83a 1 UDP 2130706431 49.212.184.** 13738 typ host
a=candidate:H31d4b83a 2 UDP 2130706430 49.212.184.** 13739 typ host
a=sendrecv
m=video 11004 RTP/AVP 31 34 98 99
a=ice-ufrag:6f1c89135c54c11f737f133616711c7a
a=ice-pwd:51cb527743bb0f520f42cffa254d3760
a=candidate:H31d4b83a 1 UDP 2130706431 49.212.184.** 11004 typ host
a=candidate:H31d4b83a 2 UDP 2130706430 49.212.184.** 11005 typ host
a=rtpmap:31 H261/90000
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;CIF4=1;VGA=1;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:98 h263-1998/90000
a=fmtp:98 F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:99 H264/90000
a=fmtp:99 redundant-pic-cap=0;parameter-add=0;packetization-mode=0;level-asymmetry-allowed=0
a=sendrecv 

You cut out too much of the SIP and I couldn’t fully reconstitute the flows from your annotations, but it looks like one of the X-Lites is not happy with using video, even though it supports compatible codecs.

a X-Lite doesn’t offer video
b X-Lite doesn’t offer video
c Asterisk offers video
d X-Lite rejects video, but says it supports all the offered video formats (camera disabled?)
e Asterisk doesn’t offer video
f X-Lite offers video, H.263 and H.263-1998
g Asterisk offers same two video formats
h X-Lite offers quite a few video formats.
g Asterisk offers four of those formats

HI David55

Thanks for your kindly support.
Thanks to your support.I have learened a lot about asterisk.

My full log is here.
ow.ly/eODrX

somehow X-lite rejects video on part D)…
(There is no problem on each camera,because exchange caller and answerer PC, caller can watch the video)

Is there anything I can do for resolving the issue?
I will try to change the clients.

Hi David55

In result,I couldn’t establish both video calls between X-lite though,
I could make work around.

This issue is something because of codecs.

SO I tried other clients.

a) Jitsi

Jitsi supports H264,so it works well.

b)LinPhone

I tried LinPhone too.
I could make video work but somehow video quality is very low.
it supports VP8,MP4V-ES,H263.
I am not sure which codec were it used.

When I tried to use VP8 only ,it doesn’t work.

I need to research and find why VP8 is not working on asterisk 11 (I want to use sipml5) though,

For now, I have found work around way.

Thanks a lot.

Asterisk 11, nor any version of Asterisk, supports pass-through for VP8.

Cheers

But my asterisk 11.0.0(patched sipml5.googlecode.com/svn/trunk/ … 3330.patch)
shows these codecs

core show codecs
.
.
200001 video h261 (H.261 Video)
200002 video h263 (H.263 Video)
200003 video h263p (H.263+ Video)
200004 video h264 (H.264 Video)
200005 video mpeg4 (MPEG4 Video)
200006 video vp8 (VP8 Video)
400001 text red (T.140 Realtime Text with redundancy)
400002 text t140 (Passthrough T.140 Realtime Text)
100013 audio siren7 (ITU G.722.1 (Siren7, licensed from Poly
.
.

BTW,
on Asterisk SVN-trunk-r373330 (patched the same patch sipml5.googlecode.com/svn/trunk/ … 3330.patch)

VP8 doesnt appear in codecs list

mmmmmmaybe you should talk to the people that wrote that patch?

Ah yes
I might have misunderstood.

I thought that the patch is for websocket support and asterisk 11.0.0 itself supports VP8.

However,In fact,this patch is for both websocket and VP8.
asterisk 11.0.0 itself doesn’t support nor websocket neither VP8.

That’s not accurate.

Asterisk 11 does support websockets over SIP. The patch you got from the sipml5 people isn’t necessary for WebSockets support in current versions of Chrome. It was necessary until Google added proper ICE support, but they’ve already done that. So now, that patch from sipml5 isn’t necessary for WebSockets. What they do have in their patch, though, is pass-through support for VP8, which no version of Asterisk has.

Thanks malcomd

I understood.
I will search VP8 patch and try Asterisk 11.0.1 with new patch.