ConfBridge doesnt make video!

tuananh141296 · December 5, 2015, 5:47pm

Hi,
Im using asterisk 13.6 on ubuntu 14.04. And now i try to make the video conference but only audio and not video. i use wireshark to check the packet, the peers do not sent video packet although i started the camera.

This is my configure :

confBridge.conf :

[default_user]
type = user
jitterbuffer = no

[default_bridge]
type = bridge
video_mode = follow_talker

extensions.conf :

exten => 200,1,ConfBridge(200)

sip debug when i start the video from the peer :

<— SIP read from UDP:192.168.0.107:5060 —>
INVITE sip:200@192.168.0.105:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK.~WCVP2ow8;rport
From: sip:1000@192.168.0.105;tag=nx4tdDqLf
To: sip:200@192.168.0.105;tag=as6a3129fb
CSeq: 25 INVITE
Call-ID: G2b5OHiRLY
Max-Forwards: 70
Subject: Media change
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 508
Contact: sip:1000@192.168.0.107;expires=3600;+sip.instance="urn:uuid:d9d2b475-1af2-4b65-b062-394357960466"
User-Agent: Linphone/3.9.1 (belle-sip/1.4.2)
Authorization: Digest realm=“asterisk”, nonce=“4ee8d14f”, algorithm=MD5, username=“1000”, uri=“sip:200@192.168.0.105:5060”, response=“18ba96f32c53d1d9a2d15d3dc134f29e”

v=0
o=1000 1536 1098 IN IP4 192.168.0.107
s=Talk
c=IN IP4 192.168.0.107
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7078 RTP/AVP 96 97 98 99 0 8 101 100 102
a=rtpmap:96 opus/48000/2
a=fmtp:96 useinbandfec=1
a=rtpmap:97 SILK/16000
a=rtpmap:98 speex/16000
a=fmtp:98 vbr=on
a=rtpmap:99 speex/8000
a=fmtp:99 vbr=on
a=rtpmap:101 telephone-event/48000
a=rtpmap:100 telephone-event/16000
a=rtpmap:102 telephone-event/8000
m=video 0 RTP/AVP 0
a=inactive
<------------->
— (14 headers 19 lines) —
Sending to 192.168.0.107:5060 (no NAT)
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found RTP audio format 100
Found RTP audio format 102
Found audio description format opus for ID 96
Found unknown media description format SILK for ID 97
Found audio description format speex for ID 98
Found audio description format speex for ID 99
Found unknown media description format telephone-event for ID 101
Found unknown media description format telephone-event for ID 100
Found audio description format telephone-event for ID 102
Capabilities: us - (ulaw|alaw|gsm|g726|vp8), peer - audio=(ulaw|alaw|opus|speex16|speex)/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 192.168.0.107:7078
Peer doesn’t provide video

<— Transmitting (no NAT) to 192.168.0.107:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK.~WCVP2ow8;received=192.168.0.107;rport=5060
From: sip:1000@192.168.0.105;tag=nx4tdDqLf
To: sip:200@192.168.0.105;tag=as6a3129fb
Call-ID: G2b5OHiRLY
CSeq: 25 INVITE
Server: FPBX-12.0.76.2(13.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:200@192.168.0.105:5060
Content-Length: 0

<------------>
Audio is at 16370
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g726 to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.0.107:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.107:5060;branch=z9hG4bK.~WCVP2ow8;received=192.168.0.107;rport=5060
From: sip:1000@192.168.0.105;tag=nx4tdDqLf
To: sip:200@192.168.0.105;tag=as6a3129fb
Call-ID: G2b5OHiRLY
CSeq: 25 INVITE
Server: FPBX-12.0.76.2(13.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:200@192.168.0.105:5060
Content-Type: application/sdp
Content-Length: 338

v=0
o=root 547098797 547098801 IN IP4 192.168.0.105
s=Asterisk PBX 13.6.0
c=IN IP4 192.168.0.105
t=0 0
m=audio 16370 RTP/AVP 0 8 3 111 102
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:102 telephone-event/8000
a=fmtp:102 0-16
a=maxptime:150
a=sendrecv
m=video 0 RTP/AVP 0

<------------>

<— SIP read from UDP:192.168.0.107:5060 —>
ACK sip:200@192.168.0.105:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5060;rport;branch=z9hG4bK.tInQs1B~8
From: sip:1000@192.168.0.105;tag=nx4tdDqLf
To: sip:200@192.168.0.105;tag=as6a3129fb
CSeq: 25 ACK
Call-ID: G2b5OHiRLY
Max-Forwards: 70
Authorization: Digest realm=“asterisk”, nonce=“4ee8d14f”, algorithm=MD5, username=“1000”, uri=“sip:200@192.168.0.105:5060”, response=“18ba96f32c53d1d9a2d15d3dc134f29e”

<------------->

as you see, no video on SDP @@ !

thanks for your help !
Ps : the video call between 2 peers works ok !

tuananh141296 · December 6, 2015, 3:48pm

Nobody cant help me ? !

david55 · December 6, 2015, 4:30pm

24 hours isn’t up yet and it is a weekend.

tuananh141296 · December 6, 2015, 5:21pm

im sorry, i forgot that today is sunday . I just want to fix it as fast as possible.

tuananh141296 · December 9, 2015, 9:49am

no idea ??

david55 · December 9, 2015, 10:01am

I’m not sure if any of the regulars use video conferencing.

ambiorixg12 · December 9, 2015, 9:00pm

Video conferencing is still on experimental phase, I think There is nothing you can do at this time.

[quote]
ConfBridge also supports basic video conferencing, though this feature is currently considered experimental.[/quote]

asterisk.org/get-started/app … conference

tuananh141296 · December 10, 2015, 4:48am

thanks for your answer. I try an other way to use video conference.
Use mcuWeb + video mixer server + asterisk.

i i add this to my conf :

extension_additional.conf:

[from-sip-external]
exten => 300,1,Dial(SIP/${EXTEN}@mcuWeb,)

sip_additional.conf

[mcuWeb]
videosupport=yes
type=peer
host=127.0.0.1 ; asterisk and mcuweb in the same server
port=5080 ; mcuWeb use this port for sip tcp/udp
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=speex
allow=h263
allow=h263p
allow=h264

when i invite the peer from mcuWeb, asterisk does not route the call to the phone.
I dont know why and trying to fix it.

ASTERISK SIP LOG :

<— SIP read from UDP:127.0.0.1:5080 —>
INVITE sip:1001@192.168.0.105:5060 SIP/2.0
Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1
CSeq: 1 INVITE
From: sip:300@mcuWeb;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5
To: sip:1001@192.168.0.105:5060
Max-Forwards: 70
User-Agent: Mobicents Sip Servlets 3.0.0-SNAPSHOT
Contact: sip:300@127.0.0.1:5080
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_468a5452-d53d-4db6-aa44-046bcfe0ef06
X-Conference-ID: 300-90fea50b-0225-44bd-91eb-dd0767147ee7
X-Participant-ID: 2
X-Participant-Token: 72e25ebe-ca9f-4f02-9ef6-2e10c7481f6e
X-Conference-Mixer-PartID: 501
X-Conference-Mixer-ID: 1628241926
Allow: INVITE,ACK,CANCEL,UPDATE,INFO,OPTIONS,BYE,INVITE,ACK,CANCEL,UPDATE,INFO,OPTIONS,BYE
Supported: timer
Session-Expires: 300;refresher=uas
Content-Type: application/sdp
Content-Length: 1435

v=0
o=- 2 1449722732663 IN IP4 192.168.0.105
s=MediaMixerSession
c=IN IP4 192.168.0.105
t=0 0
m=audio 62718 RTP/AVP 98 117 3 9 0 8
a=sendrecv
a=rtcp-mux
a=mid:audio
a=rtpmap:98 opus/48000/2
a=fmtp:98 ptime=20; maxptime=20; minptime=20; useinbandfec=0; usedtx=0; stereo=0
a=rtpmap:117 SPEEX/16000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
m=video 51280 RTP/AVP 107 99 103 104 34 109 108 110
a=sendrecv
a=rtcp-mux
a=mid:video
a=rtpmap:107 VP8/90000
a=rtcp-fb:107 nack pli
a=rtcp-fb:107 ccm fir
a=rtcp-fb:107 goog-remb
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801f
a=rtcp-fb:99 nack pli
a=rtcp-fb:99 ccm fir
a=rtcp-fb:99 goog-remb
a=rtpmap:103 H263-1998/90000
a=rtcp-fb:103 nack pli
a=rtcp-fb:103 ccm fir
a=rtcp-fb:103 goog-remb
a=rtpmap:104 MP4V-ES/90000
a=rtcp-fb:104 nack pli
a=rtcp-fb:104 ccm fir
a=rtcp-fb:104 goog-remb
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=1; QCIF=1
a=rtcp-fb:34 nack pli
a=rtcp-fb:34 ccm fir
a=rtcp-fb:34 goog-remb
a=rtpmap:109 red/90000
a=rtpmap:108 ulpfec/90000
a=rtpmap:110 rtx/90000
a=fmtp:110 apt=107
a=extmap:3 webrtc.org/experiments/rtp-h … -send-time
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
m=text 56572 RTP/AVP 105 106
a=sendrecv
a=rtcp-mux
a=mid:text
a=rtpmap:105 RED/1000
a=fmtp:105 106/106/106
a=rtpmap:106 T140/1000
<------------->
— (19 headers 56 lines) —
Sending to 127.0.0.1:5080 (no NAT)
Sending to 127.0.0.1:5080 (no NAT)
Using INVITE request as basis request - ee86cef345413250ae97c0e25e48a4e4@127.0.0.1
Found peer ‘mcuWeb’ for ‘300’ from 127.0.0.1:5080
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 98
Found RTP audio format 117
Found RTP audio format 3
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found audio description format opus for ID 98
Found audio description format SPEEX for ID 117
Found audio description format GSM for ID 3
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found RTP video format 107
Found RTP video format 99
Found RTP video format 103
Found RTP video format 104
Found RTP video format 34
Found RTP video format 109
Found RTP video format 108
Found RTP video format 110
Found video description format VP8 for ID 107
Found video description format H264 for ID 99
Found video description format H263-1998 for ID 103
Found video description format MP4V-ES for ID 104
Found video description format H263 for ID 34
Found RTP text format 105
Found RTP text format 106
Capabilities: us - (ulaw|alaw|speex|h263|h263p|h264), peer - audio=(ulaw|gsm|alaw|g722|opus|speex16)/video=(h263|h264|h263p|mpeg4|vp8|speex)/text=(red|t140), combined - (ulaw|alaw|h263|h263p|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.105:62718
Peer video RTP is at port 192.168.0.105:51280
Looking for 1001 in from-sip-external (domain 192.168.0.105)
sip_route_dump: route/path hop: sip:300@127.0.0.1:5080

<— Transmitting (no NAT) to 127.0.0.1:5080 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_468a5452-d53d-4db6-aa44-046bcfe0ef06;received=127.0.0.1
From: sip:300@mcuWeb;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5
To: sip:1001@192.168.0.105:5060
Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1
CSeq: 1 INVITE
Server: FPBX-12.0.76.2(13.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: sip:1001@127.0.0.1:5060
Content-Length: 0

<------------>
– Executing [1001@from-sip-external:1] NoOp(“SIP/mcuWeb-00000000”, “Received incoming SIP connection from unknown peer to 1001”) in new stack
– Executing [1001@from-sip-external:2] Set(“SIP/mcuWeb-00000000”, “DID=1001”) in new stack
– Executing [1001@from-sip-external:3] Goto(“SIP/mcuWeb-00000000”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/mcuWeb-00000000”, “0?checklang:noanonymous”) in new stack
– Goto (from-sip-external,s,5)
– Executing [s@from-sip-external:5] Set(“SIP/mcuWeb-00000000”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2015-12-10 11:45:47.669 ICT.
– Executing [s@from-sip-external:6] Log(“SIP/mcuWeb-00000000”, "WARNING,“Rejecting unknown SIP connection from 127.0.0.1"”) in new stack
[2015-12-10 11:45:32] WARNING[8242][C-00000000]: Ext. s:6 @ from-sip-external: “Rejecting unknown SIP connection from 127.0.0.1”
– Executing [s@from-sip-external:7] Answer(“SIP/mcuWeb-00000000”, “”) in new stack
Audio is at 16662
Video is at 127.0.0.1:14516
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h263 to SDP
Adding video codec h263p to SDP
Adding video codec h264 to SDP
Adding codec speex to SDP

<— Reliably Transmitting (no NAT) to 127.0.0.1:5080 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_468a5452-d53d-4db6-aa44-046bcfe0ef06;received=127.0.0.1
From: sip:300@mcuWeb;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5
To: sip:1001@192.168.0.105:5060;tag=as6cb0ee44
Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1
CSeq: 1 INVITE
Server: FPBX-12.0.76.2(13.6.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: sip:1001@127.0.0.1:5060
Content-Type: application/sdp
Require: timer
Content-Length: 512

v=0
o=root 1965852557 1965852557 IN IP4 127.0.0.1
s=Asterisk PBX 13.6.0
c=IN IP4 127.0.0.1
b=CT:384
t=0 0
m=audio 16662 RTP/AVP 0 8 110
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=maxptime:60
a=sendrecv
m=video 14516 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=fmtp:34 SQCIF=0;QCIF=0;CIF=1;CIF4=0;CIF16=0;VGA=0;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F
a=sendrecv
m=text 0 RTP/AVP 105 106

<------------>

<— SIP read from UDP:127.0.0.1:5080 —>
ACK sip:1001@127.0.0.1:5060 SIP/2.0
Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1
CSeq: 1 ACK
From: sip:300@mcuWeb;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5
To: sip:1001@192.168.0.105:5060;tag=as6cb0ee44
Max-Forwards: 70
User-Agent: Mobicents Sip Servlets 3.0.0-SNAPSHOT
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_3f048084-74da-40b5-895f-8875c8cf4392
Content-Length: 0

<------------->
— (9 headers 0 lines) —
[2015-12-10 11:45:32] WARNING[8242][C-00000000]: res_rtp_asterisk.c:4350 ast_rtp_read: RTP Read too short
[2015-12-10 11:45:32] WARNING[8242][C-00000000]: res_rtp_asterisk.c:4350 ast_rtp_read: RTP Read too short
> 0x7f451400b390 – Probation passed - setting RTP source address to 127.0.0.1:62718
– Executing [s@from-sip-external:8] Wait(“SIP/mcuWeb-00000000”, “2”) in new stack
> 0x7f451400df80 – Probation passed - setting RTP source address to 127.0.0.1:51280

<— SIP read from UDP:192.168.0.107:5060 —>

<------------->
– Executing [s@from-sip-external:9] Playback(“SIP/mcuWeb-00000000”, “ss-noservice”) in new stack
– <SIP/mcuWeb-00000000> Playing ‘ss-noservice.gsm’ (language ‘en’)
– Executing [s@from-sip-external:10] PlayTones(“SIP/mcuWeb-00000000”, “congestion”) in new stack
– Executing [s@from-sip-external:11] Congestion(“SIP/mcuWeb-00000000”, “5”) in new stack

<— SIP read from UDP:192.168.0.107:5060 —>

<------------->
== Spawn extension (from-sip-external, s, 11) exited non-zero on ‘SIP/mcuWeb-00000000’
– Executing [h@from-sip-external:1] Hangup(“SIP/mcuWeb-00000000”, “”) in new stack
== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/mcuWeb-00000000’
Scheduling destruction of SIP dialog ‘ee86cef345413250ae97c0e25e48a4e4@127.0.0.1’ in 32000 ms (Method: ACK)
set_destination: Parsing sip:300@127.0.0.1:5080 for address/port to send to
set_destination: set destination to 127.0.0.1:5080
Reliably Transmitting (no NAT) to 127.0.0.1:5080:
BYE sip:300@127.0.0.1:5080 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK05bd705a
Max-Forwards: 70
From: sip:1001@192.168.0.105:5060;tag=as6cb0ee44
To: sip:300@mcuWeb;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5
Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1
CSeq: 102 BYE
User-Agent: FPBX-12.0.76.2(13.6.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:127.0.0.1:5080 —>
SIP/2.0 200 Ok
CSeq: 102 BYE
Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1
From: sip:1001@192.168.0.105:5060;tag=as6cb0ee44
To: sip:300@mcuWeb;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK05bd705a
Server: Mobicents Sip Servlets 3.0.0-SNAPSHOT
Content-Length: 0

<------------->
— (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘ee86cef345413250ae97c0e25e48a4e4@127.0.0.1’ Method: ACK

MCUWEB SIP LOG :

SEVERE: Partipant 2 change state from CREATED to CONNECTING
2015-12-10 11:45:32,670 INFO [SIPTransactionStack] (http-bio-8090-exec-5) <message
from=“127.0.0.1:5080”
to=“192.168.0.105:5060”
time=“1449722732666”
isSender=“true”
transactionId=“z9hg4bkd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_468a5452-d53d-4db6-aa44-046bcfe0ef06”
callId="ee86cef345413250ae97c0e25e48a4e4@127.0.0.1"
firstLine=“INVITE sip:1001@192.168.0.105:5060 SIP/2.0”

<![CDATA[INVITE sip:1001@192.168.0.105:5060 SIP/2.0 Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1 CSeq: 1 INVITE From: ;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5 To: Max-Forwards: 70 User-Agent: Mobicents Sip Servlets 3.0.0-SNAPSHOT Contact: Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_468a5452-d53d-4db6-aa44-046bcfe0ef06 X-Conference-ID: 300-90fea50b-0225-44bd-91eb-dd0767147ee7 X-Participant-ID: 2 X-Participant-Token: 72e25ebe-ca9f-4f02-9ef6-2e10c7481f6e X-Conference-Mixer-PartID: 501 X-Conference-Mixer-ID: 1628241926 Allow: INVITE,ACK,CANCEL,UPDATE,INFO,OPTIONS,BYE,INVITE,ACK,CANCEL,UPDATE,INFO,OPTIONS,BYE Supported: timer Session-Expires: 300;refresher=uas Content-Type: application/sdp Content-Length: 1435 v=0 o=- 2 1449722732663 IN IP4 192.168.0.105 s=MediaMixerSession c=IN IP4 192.168.0.105 t=0 0 m=audio 62718 RTP/AVP 98 117 3 9 0 8 a=sendrecv a=rtcp-mux a=mid:audio a=rtpmap:98 opus/48000/2 a=fmtp:98 ptime=20; maxptime=20; minptime=20; useinbandfec=0; usedtx=0; stereo=0 a=rtpmap:117 SPEEX/16000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level m=video 51280 RTP/AVP 107 99 103 104 34 109 108 110 a=sendrecv a=rtcp-mux a=mid:video a=rtpmap:107 VP8/90000 a=rtcp-fb:107 nack pli a=rtcp-fb:107 ccm fir a=rtcp-fb:107 goog-remb a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801f a=rtcp-fb:99 nack pli a=rtcp-fb:99 ccm fir a=rtcp-fb:99 goog-remb a=rtpmap:103 H263-1998/90000 a=rtcp-fb:103 nack pli a=rtcp-fb:103 ccm fir a=rtcp-fb:103 goog-remb a=rtpmap:104 MP4V-ES/90000 a=rtcp-fb:104 nack pli a=rtcp-fb:104 ccm fir a=rtcp-fb:104 goog-remb a=rtpmap:34 H263/90000 a=fmtp:34 CIF=1; QCIF=1 a=rtcp-fb:34 nack pli a=rtcp-fb:34 ccm fir a=rtcp-fb:34 goog-remb a=rtpmap:109 red/90000 a=rtpmap:108 ulpfec/90000 a=rtpmap:110 rtx/90000 a=fmtp:110 apt=107 a=extmap:3 [webrtc.org/experiments/rtp-h ... -send-time](http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time) a=extmap:2 urn:ietf:params:rtp-hdrext:toffset m=text 56572 RTP/AVP 105 106 a=sendrecv a=rtcp-mux a=mid:text a=rtpmap:105 RED/1000 a=fmtp:105 106/106/106 a=rtpmap:106 T140/1000 ]]>

2015-12-10 11:45:32,673 INFO [SIPTransactionStack] (Mobicents-SIP-Servlets-UDPMessageChannelThread-16) <message
from=“192.168.0.105:5060”
to=“127.0.0.1:5080”
time=“1449722732672”
isSender=“false”
transactionId=“z9hg4bkd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_468a5452-d53d-4db6-aa44-046bcfe0ef06”
callId="ee86cef345413250ae97c0e25e48a4e4@127.0.0.1"
firstLine=“SIP/2.0 100 Trying”

<![CDATA[SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_468a5452-d53d-4db6-aa44-046bcfe0ef06;received=127.0.0.1 From: ;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5 To: Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1 CSeq: 1 INVITE Server: FPBX-12.0.76.2(13.6.0) Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer Session-Expires: 300;refresher=uas Contact: Content-Length: 0 ]]>

2015-12-10 11:45:32,681 INFO [SIPTransactionStack] (Mobicents-SIP-Servlets-UDPMessageChannelThread-17) <message
from=“192.168.0.105:5060”
to=“127.0.0.1:5080”
time=“1449722732681”
isSender=“false”
transactionId=“z9hg4bkd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_468a5452-d53d-4db6-aa44-046bcfe0ef06”
callId="ee86cef345413250ae97c0e25e48a4e4@127.0.0.1"
firstLine=“SIP/2.0 200 OK”

<![CDATA[SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_468a5452-d53d-4db6-aa44-046bcfe0ef06;received=127.0.0.1 From: ;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5 To: ;tag=as6cb0ee44 Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1 CSeq: 1 INVITE Server: FPBX-12.0.76.2(13.6.0) Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE Supported: replaces,timer Session-Expires: 300;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 512 v=0 o=root 1965852557 1965852557 IN IP4 127.0.0.1 s=Asterisk PBX 13.6.0 c=IN IP4 127.0.0.1 b=CT:384 t=0 0 m=audio 16662 RTP/AVP 0 8 110 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 speex/8000 a=maxptime:60 a=sendrecv m=video 14516 RTP/AVP 34 103 99 a=rtpmap:34 H263/90000 a=fmtp:34 SQCIF=0;QCIF=0;CIF=1;CIF4=0;CIF16=0;VGA=0;F=0;I=0;J=0;T=0;K=0;N=0;BPP=0;HRD=0 a=rtpmap:103 h263-1998/90000 a=rtpmap:99 H264/90000 a=fmtp:99 profile-level-id=42801F a=sendrecv m=text 0 RTP/AVP 105 106 ]]>

Dec 10, 2015 11:45:32 AM org.murillo.mcuWeb.RTPParticipant doInvite
WARNING: doInvite [idSession:(18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5;ee86cef345413250ae97c0e25e48a4e4@127.0.0.1;d4ad8764-0848-4db6-85aa-6a14058ef3e5;mcuWeb)]
Dec 10, 2015 11:45:32 AM org.murillo.mcuWeb.RTPParticipant onSDPNegotiationDone
INFO: onSDPNegotiationDone videoBitrate:320
Dec 10, 2015 11:45:32 AM org.murillo.MediaServer.XmlRpcMcuClient SetRTPProperties
INFO: SetRTPProperties(1,628,241,926,501,0,{})
Dec 10, 2015 11:45:32 AM org.murillo.MediaServer.XmlRpcMcuClient SetRTPProperties
INFO: SetRTPProperties(1,628,241,926,501,1,{useFEC=1})
2015-12-10 11:45:32,716 INFO [SipServletResponseImpl] (Mobicents-SIP-Servlets-UDPMessageChannelThread-17) ackRequest just created ACK sip:1001@127.0.0.1:5060 SIP/2.0
Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1
CSeq: 1 ACK
From: sip:300@mcuWeb;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5
To: sip:1001@192.168.0.105:5060;tag=as6cb0ee44
Max-Forwards: 70
User-Agent: Mobicents Sip Servlets 3.0.0-SNAPSHOT
Content-Length: 0

2015-12-10 11:45:32,718 INFO [SIPTransactionStack] (Mobicents-SIP-Servlets-UDPMessageChannelThread-17) <message
from=“127.0.0.1:5080”
to=“127.0.0.1:5060”
time=“1449722732717”
isSender=“true”
transactionId=“z9hg4bkd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_3f048084-74da-40b5-895f-8875c8cf4392”
callId="ee86cef345413250ae97c0e25e48a4e4@127.0.0.1"
firstLine=“ACK sip:1001@127.0.0.1:5060 SIP/2.0”

<![CDATA[ACK sip:1001@127.0.0.1:5060 SIP/2.0 Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1 CSeq: 1 ACK From: ;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5 To: ;tag=as6cb0ee44 Max-Forwards: 70 User-Agent: Mobicents Sip Servlets 3.0.0-SNAPSHOT Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKd4ad8764-0848-4db6-85aa-6a14058ef3e5_655e17f9_3f048084-74da-40b5-895f-8875c8cf4392 Content-Length: 0 ]]>

Dec 10, 2015 11:45:32 AM org.murillo.mcuWeb.Participant setState
SEVERE: Partipant 2 change state from CONNECTING to CONNECTED
Dec 10, 2015 11:45:32 AM org.murillo.MediaServer.XmlRpcMcuClient AddMosaicParticipant
INFO: AddMosaicParticipant(1,628,241,926,0,501,{3})
Dec 10, 2015 11:45:32 AM org.murillo.MediaServer.XmlRpcMcuClient AddSidebarParticipant
INFO: AddSidebarParticipant(1,628,241,926,0,501)
Dec 10, 2015 11:45:32 AM org.murillo.MediaServer.XmlRpcMcuClient SetAudioCodec
INFO: SetAudioCodec(1,628,241,926,501,0)
Dec 10, 2015 11:45:32 AM org.murillo.MediaServer.XmlRpcMcuClient StartSending
INFO: StartSending(1,628,241,926,501,0,127.0.0.1,16,662,{0=0, 8=8})
Dec 10, 2015 11:45:32 AM org.murillo.MediaServer.XmlRpcMcuClient SetVideoCodec
INFO: SetVideoCodec(1,628,241,926,501,99,0,10,320,100,{h264.profile-level-id=42801f})
Dec 10, 2015 11:45:32 AM org.murillo.MediaServer.XmlRpcMcuClient StartSending
INFO: StartSending(1,628,241,926,501,1,127.0.0.1,14,516,{34=34, 99=99, 103=103})
2015-12-10 11:45:45,007 INFO [SIPTransactionStack] (Mobicents-SIP-Servlets-UDPMessageChannelThread-18) <message
from=“127.0.0.1:5060”
to=“127.0.0.1:5080”
time=“1449722745005”
isSender=“false”
transactionId=“z9hg4bk05bd705a”
callId="ee86cef345413250ae97c0e25e48a4e4@127.0.0.1"
firstLine=“BYE sip:300@127.0.0.1:5080 SIP/2.0”

<![CDATA[BYE sip:300@127.0.0.1:5080 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK05bd705a Max-Forwards: 70 From: ;tag=as6cb0ee44 To: ;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5 Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1 CSeq: 102 BYE User-Agent: FPBX-12.0.76.2(13.6.0) X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 ]]>

2015-12-10 11:45:45,012 INFO [SIPTransactionStack] (Mobicents-SIP-Servlets-UDPMessageChannelThread-18) <message
from=“127.0.0.1:5080”
to=“127.0.0.1:5060”
time=“1449722745011”
isSender=“true”
transactionId=“z9hg4bk05bd705a”
callId="ee86cef345413250ae97c0e25e48a4e4@127.0.0.1"
firstLine=“SIP/2.0 200 Ok”

<![CDATA[SIP/2.0 200 Ok CSeq: 102 BYE Call-ID: ee86cef345413250ae97c0e25e48a4e4@127.0.0.1 From: ;tag=as6cb0ee44 To: ;tag=18784397_6ee792b5_655e17f9_d4ad8764-0848-4db6-85aa-6a14058ef3e5 Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK05bd705a Server: Mobicents Sip Servlets 3.0.0-SNAPSHOT Content-Length: 0 Thanks for your help again !

david55 · December 10, 2015, 7:41am

A video call was set up on the incoming side. The dialplan rejected it. Please consult the people who wrote the dialplan (it looks like it is some sort of GUI dialplan; no GUIs are supported here.)

This sort of rejection is typical for a call to FreePBX from an unknown peer.

tuananh141296 · December 10, 2015, 8:52am

You are right ! i installed Freepbx with asterisk, so i need to allow asterisk to handle the invite message which send by anonymous in freepbx.
That make it work for me now.

Thanks for your hint alot .

One more question. Like i describe above.
i add this in my extensions_additional.conf file :

[from-sip-external]
exten => 300,1,Dial(SIP/${EXTEN}@mcuWeb,)

but when i call 300, asterisk did not send any message to mcuWeb (it didnt route the call) although mcuWeb can call to an other peers !