[asterisk-users] Video doesn't work for outgoing call?

Chih-Wei Huang asterisk at cwhuang.idv.tw
Wed Sep 19 22:32:53 CDT 2007


I've tried to put a call file to /var/spool/asterisk/outgoing/
to make an outgoing video call, but not succeeded.
I could hear the audio, but no video.

The asterisk version is 1.4.10, with videosupport=yes
The client is eyebeam 1.5.7, with h263 support.

Here are some debug messages.
It shows the client and asterisk negotiated the video capabilities
without problem. However, the 'show channel ...' command
shows there is only audio channel, no video channel.
I don't understand why.

The file I tried to play is
$ ls -al /var/lib/asterisk/VOD/jolin-512k*
-rw-r--r-- 1 root root   7975341 2007-03-09 13:16 
/var/lib/asterisk/VOD/jolin-512k.gsm
-rw-r--r-- 1 root root 236916736 2007-03-09 13:16 
/var/lib/asterisk/VOD/jolin-512k.h263

both .gsm and .h263 are available.

I'm sure the media files have no problem,
since I can see the video by making a call from
eyebeam to asterisk.

I have also tried to make outgoing call by the manager API.
There is no video, either.

Any idea why video doesn't work for outgoing call from asterisk?


cwhuang*CLI> sip set debug peer 403
SIP Debugging Enabled for IP: 172.16.148.129:36042
Reliably Transmitting (NAT) to 172.16.148.129:36042:
OPTIONS sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901 SIP/2.0
Via: SIP/2.0/UDP 172.16.148.1:5060;branch=z9hG4bK0dcbc40d;rport
From: "asterisk" <sip:asterisk at 172.16.148.1>;tag=as25cd8375
To: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>
Contact: <sip:asterisk at 172.16.148.1>
Call-ID: 18ce68323649d71250c5267f6fae3c15 at 172.16.148.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 20 Sep 2007 03:14:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
cwhuang*CLI>
<--- SIP read from 172.16.148.129:36042 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.148.1:5060;branch=z9hG4bK0dcbc40d;rport=5060
Contact: <sip:172.16.148.129:36042>
To: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>;tag=05435c64
From: "asterisk"<sip:asterisk at 172.16.148.1>;tag=as25cd8375
Call-ID: 18ce68323649d71250c5267f6fae3c15 at 172.16.148.1
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
SUBSCRIBE, INFO
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog 
'18ce68323649d71250c5267f6fae3c15 at 172.16.148.1' Method: OPTIONS
cwhuang*CLI>
<--- SIP read from 172.16.148.129:36042 --->



<------------->
--- (0 headers 1 lines) ---
     -- Attempting call on SIP/403 for s at broadcast:1 (Retry 1)
Video is at 172.16.148.1 port 18182
Audio is at 172.16.148.1 port 18108
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x80000 (h263) to SDP
Adding codec 0x100000 (h263p) to SDP
Adding codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.16.148.129:36042:
INVITE sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901 SIP/2.0
Via: SIP/2.0/UDP 172.16.148.1:5060;branch=z9hG4bK1d87664a;rport
From: "555" <sip:555 at 172.16.148.1>;tag=as74d0c1cf
To: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>
Contact: <sip:555 at 172.16.148.1>
Call-ID: 2ecf0d35433f334e429760f65c4ebb91 at 172.16.148.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 20 Sep 2007 03:15:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 441

v=0
o=root 7226 7226 IN IP4 172.16.148.1
s=session
c=IN IP4 172.16.148.1
b=CT:384
t=0 0
m=audio 18108 RTP/AVP 18 3 0 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 18182 RTP/AVP 34 103 99
a=rtpmap:34 H263/90000
a=rtpmap:103 h263-1998/90000
a=rtpmap:99 H264/90000
a=sendrecv

---
cwhuang*CLI>
<--- SIP read from 172.16.148.129:36042 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.148.1:5060;branch=z9hG4bK1d87664a;rport=5060
Contact: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>
To: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>;tag=d864d761
From: "555"<sip:555 at 172.16.148.1>;tag=as74d0c1cf
Call-ID: 2ecf0d35433f334e429760f65c4ebb91 at 172.16.148.1
CSeq: 102 INVITE
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
cwhuang*CLI>
<--- SIP read from 172.16.148.129:36042 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.148.1:5060;branch=z9hG4bK1d87664a;rport=5060
Contact: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>
To: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>;tag=d864d761
From: "555"<sip:555 at 172.16.148.1>;tag=as74d0c1cf
Call-ID: 2ecf0d35433f334e429760f65c4ebb91 at 172.16.148.1
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, 
SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 569

v=0
o=- 9 2 IN IP4 172.16.148.129
s=CounterPath eyeBeam 1.5
c=IN IP4 172.16.148.129
t=0 0
m=audio 18342 RTP/AVP 18 3 0 101
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:07F885091DAF4305868C0F432F6512CF
m=video 35840 RTP/AVP 34 115 125
a=fmtp:34 QCIF=2 MAXBR=1960
a=fmtp:115 QCIF=2 MAXBR=1960
a=fmtp:125 profile-level-id=42e015; max-br=4000; max-mbps=19800
a=rtpmap:34 H263/90000
a=rtpmap:115 H263-1998/90000
a=rtpmap:125 H264/90000
a=sendrecv
a=x-rtp-session-id:399907E52958448ABF8B7176DCD44BD5

<------------->
--- (11 headers 20 lines) ---
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 101
Found RTP video format 34
Found RTP video format 115
Found RTP video format 125
Peer audio RTP is at port 172.16.148.129:18342
Found description format telephone-event for ID 101
Found description format H263 for ID 34
Found description format H263-1998 for ID 115
Found description format H264 for ID 125
Capabilities: us - 0x380106 (gsm|ulaw|g729|h263|h263p|h264), peer - 
audio=0x380106 (gsm|ulaw|g729|h263|h263p|h264)/video=0x380000 
(h263|h263p|h264), combined - 0x380106 (gsm|ulaw|g729|h263|h263p|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.148.129:18342             <=== audio OK
Peer video RTP is at port 172.16.148.129:35840             <=== video OK
list_route: hop: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>
set_destination: Parsing 
<sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901> for 
address/port to send to
set_destination: set destination to 172.16.148.129, port 36042
Transmitting (NAT) to 172.16.148.129:36042:
ACK sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901 SIP/2.0
Via: SIP/2.0/UDP 172.16.148.1:5060;branch=z9hG4bK2aa30420;rport
From: "555" <sip:555 at 172.16.148.1>;tag=as74d0c1cf
To: <sip:403 at 172.16.148.129:36042;rinstance=24208107cb163901>;tag=d864d761
Contact: <sip:555 at 172.16.148.1>
Call-ID: 2ecf0d35433f334e429760f65c4ebb91 at 172.16.148.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
     -- Executing [s at broadcast:1] Goto("SIP/403-097cc8e8", 
"customer|701|1") in new stack
     -- Goto (customer,701,1)
     -- Executing [701 at customer:1] Goto("SIP/403-097cc8e8", "vod|701|1") 
in new stack
     -- Goto (vod,701,1)
     -- Executing [701 at vod:1] Set("SIP/403-097cc8e8", 
"MOIVE=jolin-512k") in new stack
     -- Executing [701 at vod:2] Ringing("SIP/403-097cc8e8", "") in new stack
     -- Executing [701 at vod:3] Goto("SIP/403-097cc8e8", "t|1") in new stack
     -- Goto (vod,t,1)
     -- Executing [t at vod:1] Macro("SIP/403-097cc8e8", 
"playvod|/var/lib/asterisk/VOD/jolin-512k") in new stack
     -- Executing [s at macro-playvod:1] Wait("SIP/403-097cc8e8", "1") in 
new stack
     -- Executing [s at macro-playvod:2] Set("SIP/403-097cc8e8", 
"FROM_IVR=yes") in new stack
     -- Executing [s at macro-playvod:3] Set("SIP/403-097cc8e8", 
"TIMEOUT(digit)=5") in new stack
     -- Digit timeout set to 5
     -- Executing [s at macro-playvod:4] Set("SIP/403-097cc8e8", 
"TIMEOUT(response)=5") in new stack
     -- Response timeout set to 5
     -- Executing [s at macro-playvod:5] Macro("SIP/403-097cc8e8", 
"play|/var/lib/asterisk/VOD/jolin-512k") in new stack
     -- Executing [s at macro-play:1] Set("SIP/403-097cc8e8", 
"LANGUAGE()=zh") in new stack
     -- Executing [s at macro-play:2] Answer("SIP/403-097cc8e8", "1000") in 
new stack
     -- Executing [s at macro-play:3] BackGround("SIP/403-097cc8e8", 
"/var/lib/asterisk/VOD/jolin-512k") in new stack
     -- <SIP/403-097cc8e8> Playing '/var/lib/asterisk/VOD/jolin-512k' 
(language 'zh')
cwhuang*CLI>
<--- SIP read from 172.16.148.129:36042 --->



<------------->
--- (0 headers 1 lines) ---
cwhuang*CLI> show ch
channel       channels      channeltypes  channeltype
cwhuang*CLI> show channel SIP/403-097cc8e8
  -- General --
            Name: SIP/403-097cc8e8
            Type: SIP
        UniqueID: 1190258125.131
       Caller ID: 555
  Caller ID Name: (N/A)
     DNID Digits: (N/A)
           State: Up (6)
           Rings: 0
   NativeFormats: 0x4 (ulaw)
     WriteFormat: 0x2 (gsm)
      ReadFormat: 0x100 (g729)
  WriteTranscode: Yes
   ReadTranscode: Yes
1st File Descriptor: 38
       Frames in: 494
      Frames out: 510
  Time to Hangup: 0
    Elapsed Time: 0h0m13s
   Direct Bridge: <none>
Indirect Bridge: <none>
  --   PBX   --
         Context: macro-play
       Extension: s
        Priority: 3
      Call Group: 0
    Pickup Group: 0
     Application: BackGround
            Data: /var/lib/asterisk/VOD/jolin-512k
     Blocking in: ast_waitfor_nandfds
       Variables:
MACRO_DEPTH=2
ARG1=/var/lib/asterisk/VOD/jolin-512k
MACRO_PRIORITY=5
MACRO_CONTEXT=macro-playvod
MACRO_EXTEN=s
FROM_IVR=yes
MOIVE=jolin-512k
RECORDED=broadcast/msg-24
SIPCALLID=2ecf0d35433f334e429760f65c4ebb91 at 172.16.148.1

   CDR Variables:
level 1: clid=555
level 1: src=555
level 1: dst=t
level 1: dcontext=vod
level 1: channel=SIP/403-097cc8e8
level 1: lastapp=BackGround
level 1: lastdata=/var/lib/asterisk/VOD/jolin-512k
level 1: start=2007-09-20 11:15:25
level 1: answer=2007-09-20 11:15:28
level 1: end=2007-09-20 11:15:28
level 1: duration=0
level 1: billsec=0
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1190258125.131




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