[Asterisk-video] Confiance video mixer
Lorenzo Miniero
lorenzo.miniero at unina.it
Thu Sep 13 03:47:55 CDT 2007
Hi Takashi,
I'm currently abroad so I can't fully work on debugging your problem at the
moment. However, I'll be back to my lab on Monday: I'll promise I'll dive into
it as soon as I'll step in there :)
As some first hits:
1) check that the video device in the Minisip preferences is correctly setup
2) check that the videomixer settings are correctly set in the xcon.conf file
3) if so, do you at least see the Confiance logo when joining a conference?
Hear you back on Monday, best regards,
Lorenzo
--
Lorenzo Miniero, Junior Researcher
Dipartimento di Informatica e Sistemistica
Università degli Studi di Napoli "Federico II"
Via Claudio 21 -- 80125 Napoli (Italy)
Phone: +390817683821 - Fax: +390817683816
Email: lorenzo.miniero at unina.it
Scrive Takashi Ohashi <ohashi at fko.it-tokyo.co.jp>:
> 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,
>
----------------------------------------------------------------
This message was sent using IMP, the Internet Messaging Program.
More information about the asterisk-video
mailing list