[Asterisk-video] Confiance video mixer
Takashi Ohashi
ohashi at fko.it-tokyo.co.jp
Thu Sep 13 01:57:12 CDT 2007
Hi
I tryed to build CONFIANCE Video mixer and confirm video mixed,
refering to http://confiance.sourceforge.net/, especially Installer script.
I constructed the environment below.
* Asterisk(1.4.3) with CONFIANCE
patch(confiance_patch_asterisk_jun14th-07.diff)
* CONFIANCE video mixer(confiance_vm)
* minisip(revision 2760) with CONFIANCE
patch(confiance_patch_minisip_jun14th-07.diff)
But unfortunately it does not work correct.
Could someone help me with this problem?
Perhaps Lorenzo!
I show the infomation about our environment bellow.
1. operation
1.1 server
(1) startup video mixer
confiance_vm 7000
(2) startup asterisk
asterisk -gcvvvvvvvvvvv
1.2 minisip
1.2.1 startup
# ./build.pl run minisip
1.2.2 join an XCON Conference
I operate minisip GUI as shown below.
(1)enable and check XCON on "VIEW" menu bar.
(2)input Confernce number(8671000) to "Extension" input area on "XCON"
tab window.
(3)push "join this XCON Conference" button on "XCON" tab window.
Now I wonder how to use video conference on minisip, because I can't find
document about video confernece.
Please tell me correct operation about video confernece, if I mistake.
2. packet between minisip and asterisk
After I operate by the procedure above,
I confirmed the session between minisip and asterisk.
But it seemed that only audio session was established, but video session was
not.
After established session, I confirmed audio RTP packet ,but not video RTP
packet.
I show sip packet between minisip and asterisk below.
------------------SIP PACKET FLOW----------------------------------
(1) minisip(IPaddr:192.168.1.173) -> asterisk(IPaddr:192.168.1.21)
Request-Line: INVITE sip:8671000 at 192.168.1.21 SIP/2.0
Message Header
Route: <sip:192.168.1.21:5060;transport=UDP;lr>
From: <sip:2001 at 192.168.1.21>;tag=1619364546
To: <sip:8671000 at 192.168.1.21>
Call-ID: 438183866 at 192.168.1.173
CSeq: 801 INVITE
Contact: <sip:2001 at 192.168.1.173:5060;transport=UDP>;expires=1000
User-Agent: Minisip
Supported: 100rel
Content-Type: application/sdp
Via: SIP/2.0/UDP 192.168.1.173:5060;rport;branch=z9hG4bK976758295
Content-Length: 357
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 3344 3344 IN IP4 192.168.1.173
Session Name (s): Minisip Session
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 30554 RTP/AVP 0
101
Connection Information (c): IN IP4 192.168.1.173
Media Attribute (a): rtpmap:0 PCMU/8000/1
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Description, name and address (m): video 30992 RTP/AVP 105
101
Connection Information (c): IN IP4 192.168.1.173
Media Attribute (a): rtpmap:105 h263-1998/90000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): framesize:34 176-144
(2) asterisk -> minisip
Status-Line: SIP/2.0 100 Trying
(3) asterisk -> minisip
Status-Line: SIP/2.0 200 OK
Message Header
Via: SIP/2.0/UDP
192.168.1.173:5060;branch=z9hG4bK976758295;received=192.168.1.173;rport=5060
From: <sip:2001 at 192.168.1.21>;tag=1619364546
To: <sip:8671000 at 192.168.1.21>;tag=as446a6ab2
Call-ID: 438183866 at 192.168.1.173
CSeq: 801 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:8671000 at 192.168.1.21>
Content-Type: application/sdp
Content-Length: 240
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 11774 11774 IN IP4
192.168.1.21
Session Name (s): session
Connection Information (c): IN IP4 192.168.1.21
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 25836 RTP/AVP 0
101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): silenceSupp:off - - - -
Media Attribute (a): ptime:20
Media Attribute (a): sendrecv
(4) minisip -> asterisk
Request-Line: ACK sip:8671000 at 192.168.1.21 SIP/2.0
(5) asterisk -> minisip
Request-Line: INVITE sip:2001 at 192.168.1.173:5060;transport=UDP SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK3b9cc93d;rport
From: <sip:8671000 at 192.168.1.21>;tag=as446a6ab2
To: <sip:2001 at 192.168.1.21>;tag=1619364546
Contact: <sip:8671000 at 192.168.1.21>
Call-ID: 438183866 at 192.168.1.173
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 420
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 11774 11775 IN IP4
192.168.1.21
Session Name (s): session
Connection Information (c): IN IP4 192.168.1.21
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 25836 RTP/AVP 0
101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-16
Media Attribute (a): silenceSupp:off - - - -
Media Attribute (a): ptime:20
Media Attribute (a): label:10
Media Attribute (a): sendrecv
Media Description, name and address (m): application 2345
TCP/BFCP *
Media Attribute (a): setup:passive
Media Attribute (a): connection:new
Media Attribute (a): floorctrl:s-only
Media Attribute (a): confid:8671000
Media Attribute (a): userid:1
Media Attribute (a): floorid:11 m-stream:10
Media Attribute (a): floorid:22 m-stream:11
(6) minisip -> asterisk
Status-Line: SIP/2.0 200 OK
Message Header
Max-Forwards: 70
Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK3b9cc93d;rport=5060
From: <sip:8671000 at 192.168.1.21>;tag=as446a6ab2
To: <sip:2001 at 192.168.1.21>;tag=1619364546
Call-ID: 438183866 at 192.168.1.173
CSeq: 102 INVITE
Contact: <sip:2001 at 192.168.1.173:5060;transport=UDP>
Content-Type: application/sdp
Content-Length: 439
Message body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 3344 3344 IN IP4 192.168.1.173
Session Name (s): Minisip Session
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 30554 RTP/AVP 0
101
Connection Information (c): IN IP4 192.168.1.173
Media Attribute (a): rtpmap:0 PCMU/8000/1
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Description, name and address (m): video 30992 RTP/AVP 105
101
Connection Information (c): IN IP4 192.168.1.173
Media Attribute (a): rtpmap:105 h263-1998/90000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute (a): fmtp:101 0-15
Media Attribute (a): framesize:34 176-144
Media Description, name and address (m): application 9 TCP/BFCP
*
Media Attribute (a): setup:active
Media Attribute (a): floorctrl:c-only
Media Attribute (a): connection:new
(7) asterisk -> minisip
Request-Line: ACK sip:2001 at 192.168.1.173:5060;transport=UDP SIP/2.0
----------------------------------------------------------------------------
3. Asterisk console log
I show the console log on Asterisk below, when I join XCON conference on
minisip.
--------------------Asterisk console log(join XCON conference)--------------
*CLI> -- Executing [8671000 at default:1] MeetMe("SIP/2001-08f8aa20",
"8671000|B|") in new stack
== Parsing '/etc/asterisk/xcon.conf': Found
[Sep 13 10:29:33] WARNING[11809]: config.c:756 process_text_line: No '='
(equal sign) in line 52 of /etc/asterisk/xcon.conf
-- The new local conference (ConferenceID: 8671000) has been added to
the BFCP Server:
-- Floor: Audio, ID 11 (unlimited users)
-- Floor: Video, ID 22 (limited users)
-- Adding conference to the BFCP Server: DONE
-- Created XCON conference 1023 for conference '8671000'
-- Requesting new VideoMixer session for conference 8671000
-- New Participant has UserID 1 (Conference 8671000)...
-- CallerID: , URI: sip:2001 at 192.168.1.173:5060
[Sep 13 10:29:33] WARNING[11809]: app_meetme.c:2841 conf_run: Couldn't add
UserID 1 to Conference 8671000 Users' list...
-- Sending required BFCP+MSRP information to chan_sip...
-- BFCP information structure for SDP received from MeetMe...
-- Video Format: no video
-- [CVM] Conference 8671000 --> Session 1
-- <SIP/2001-08f8aa20> Playing 'conf-onlyperson' (language 'en')
-- ACK from XCON client received, requesting reinvite...
-- Transmitting pending reinvite with BFCP information...
-- Building SDP+BFCP/MSRP...
-- Actually sending reinvite with BFCP information...
-- BFCP [8671000/1/1] <--- Hello
-- BFCP [8671000/1/1] ---> HelloAck
-- BFCP [8671000/2/1] <--- FloorQuery
-- BFCP [8671000/2/1] ---> FloorStatus
-- [XCON] XconScheduler: QueryUsers
-- [XCON] InfoUsers
-- Parsing BFCP information in SIP OK's SDP: TCP/BFCP (bfcp port = 9,
discard)...
-- UserID 1 (Conference 8671000) MUTED
-- <SIP/2001-08f8aa20> Playing 'conf-muted' (language 'en')
-- Notifying AudioFloor change to chan_sip...
-- Not notifying AudioFloor change to chan_sip, user just joined...
---------------------------------------------------------------------------
Regards,
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