[Asterisk-video] R: Re: Asterisk Video->debug Confiance videomixer
Sergio Garcia Murillo
sergio.garcia at fontventa.com
Wed Apr 9 10:12:17 CDT 2008
By the way, I was thinking in implementing a multiplexor application in which you could bridge audio and video streams at will between channels.
For example it could be useful for receiving a 3g call in asterisk, call a voice agent and play a video to the caller controlled by dtmf by the agent.
3g<----------->agent
^ audio /
\ /
\video / dtmf
app_mp4 </
Would a new application be needed or it coudl be done by other means??
BR
Sergio
----- Original Message -----
From: borja.sixto at i6net.com [mailto:borja.sixto at i6net.com]
To: asterisk-video at lists.digium.com,sergio.garcia at fontventa.com
Cc: asterisk-video at lists.digium.com
Sent: Wed, 09 Apr 2008 16:00:19 +0200
Subject: Re: [Asterisk-video] R: Re: Asterisk Video->debug Confiance videomixer
Have you try app_conference.
http://sourceforge.net/projects/appconference/
There is no video mixing, but a video switching controled by the voice or by a
DTMF.
Regards,
Tech from i6net
Selon Sergio Garcia Murillo <sergio.garcia at fontventa.com>:
> You could try my multiconference solution (mcuWeb+MediaMixer)
>
> http://sip.fontventa.com/content/view/32/63/
>
> Best regards
> Sergio
>
> ----- Original Message -----
> From: marti2001 at tin.it [mailto:marti2001 at tin.it]
> To: asterisk-video at lists.digium.com
> Sent: Wed, 9 Apr 2008 12:13:50 +0100 (GMT+01:00)
> Subject: [Asterisk-video] R: Re: Asterisk_Video->debug Confiance_videomixer
>
>
> Hi!
> I don't know...my goal is to do a videoconference whith 3 users or
> more. Do you know something that do this?
> I'm using
> Confiance_videomixer..
>
> Thanks!
>
> MARTINA
> ----Messaggio originale----
> Da: josemrecio at gmail.com
> Data: 8-apr-2008 11.48 PM
> A: "Development
> discussion of video media support in Asterisk"<asterisk-video at lists.
> digium.com>
> Ogg: Re: [Asterisk-video] R: Re: Asterisk_Video->debug
> Confiance_videomixer
>
> I have never used Confiance ...
> Does the service
> work fine if the call is just between 3G and XLite?
>
> -----Mensaje
> original-----
> De: asterisk-video-bounces at lists.digium.com
> [mailto:
> asterisk-video-bounces at lists.digium.com] En nombre de
> marti2001 at tin.it
> Enviado el: martes, 08 de abril de 2008 18:56
> Para: asterisk-
> video at lists.digium.com
> Asunto: [Asterisk-video] R: Re: Asterisk_Video-
> >debug Confiance_videomixer
>
> Thanks for the mail..
> I sow the options
> of X-lite and I change samething options but the problem
> is the same!
>
> The problem can be the
> client, what do you think?
>
> Thanks adn good
> evening!
> MARTINA
>
>
> ----
> Messaggio originale----
> Da: josemrecio at gmail.
> com
> Data: 8-apr-2008 3.23
> PM
> A: "Development discussion of video media
> support in Asterisk"
> <asterisk-video at lists.digium.com>
> Ogg: Re:
> [Asterisk-video]
> Asterisk_Video->debug Confiance_videomixer
>
> Suggestion:
> Message
> "impossible bitrate constraints, this will fail"
> ... try different XLite
> bandwidth settings (Advanced -> Network
> options)
>
> -----Mensaje
> original-----
> De: asterisk-video-bounces at lists.
> digium.com
> [mailto:
> asterisk-video-bounces at lists.digium.com] En nombre
> de marti2001 at tin.it
> Enviado el: martes, 08 de abril de 2008 15:05
> Para:
> asterisk-
> video at lists.digium.com
> Asunto: [Asterisk-video]
> Asterisk_Video->debug Confiance_videomixer
>
> HI!
>
> ##Debug di CONFIANCE
> VM IF(IN sip.conf):
> allow=h263p ##
> maxcallbitrate=384
>
>
> CVM_CLI*>
> CVM_CLI*> New client
> connected
> (192.168.23.22:58070)
> CVM_CLI*> Session
> 1 created
> CVM_CLI*>
> Process
> thread started for Session n.1 (Conference
> 8671000) CVM_CLI*> Process thread
> for Session n.1 (Conference 8671000)
> put to sleep...
> ortp-
> message-Using permissive algorithm
> ___________________ Inside
> cvm_peer_new
> ________________________CVM_CLI*> Source 1 added to Session 1
> CVM_CLI*>
> Source thread: session 1, peer 1, RTP /17046 CVM_CLI*> RTP: OK
> CVM_CLI*> Context: OK CVM_CLI*>
> Decoder: OK
> CVM_CLI*> Frames: OK
> CVM_CLI*> Waking up the process
> thread...
> CVM_CLI*> Process thread for
> Session n.1 (Conference 8671000) woken up
> ortp-message-Using permissive
> algorithm ___________________ Inside
> cvm_peer_new
> ________________________CVM_CLI*> MMX support enabled
> ****************** PT is
> 103******************************************** PT is 103
> *********************************
> [h263p @ 0xb7dc3930]impossible
> bitrate constraints, this will fail
> CVM_CLI*> Destination 2 added to
> Session 1
> [h263p @ 0xb7dc3930]rc buffer underflow
>
>
>
>
>
>
>
> ##Debug di
> Asterisk##
>
>
> *CLI> [Apr 7 16:50:59] NOTICE[11321]: chan_sip.c:14761
> handle_request_subscribe: Received SIP subscribe for peer without
> mailbox: 101
> -- Executing [8671000 at default:1] Answer("SIP/101-
> 081cba58", "") in new
> stack
> -- Executing [8671000 at default:2] MeetMe
> ("SIP/101-081cba58", "8671000|B|") in new stack == Parsing
> '/etc/asterisk/xcon.conf': Found
> -- 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:
> 101, URI: sip:101 at 192.168.23.240:
> 15709
> [Apr 7 16:51:09] WARNING
> [11333]: 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...
> -- ACK from XCON client
> received,
> requesting reinvite...
> -- Transmitting pending reinvite
> with
> BFCP
> information...
> -- Building SDP+BFCP/MSRP...
> -- Actually
> sending
> reinvite with BFCP information...
> -- [CVM] Conference
> 8671000
> -->
> Session 1
> -- Started Video RTP Channel for user 1 on port 10836,
> notifying
> VideoMixer...
> -- Format 1048576 --> 103/H.
> 263+ (confirmed)
> -- Video Format: H.263+
> -- <SIP/101-081cba58>
> Playing 'conf-
> onlyperson' (language 'en')
> -- Parsing BFCP
> information in SIP OK's
> SDP: TCP/BFCP (bfcp port = 0, bound)...
> [Apr 7
> 16:51:09] NOTICE
> [11333]: rtp.c:1256 ast_rtp_read: Unknown RTP codec 126 received from
> '192.168.23.240'
> [Apr 7 16:51:09] NOTICE[11333]: rtp.c:
> 1256
> ast_rtp_read: Unknown RTP codec 126 received from '192.168.23.240'
> [Apr
> 7 16:51:09] NOTICE[11333]: rtp.c:1256 ast_rtp_read: Unknown RTP codec
> 126 received from '192.168.23.240'
> -- [CVM] User 1 (8671000)
> --
> >
> Session 1 / Peer 1
> -- Incoming H.263+ (103) Video RTP Channel waiting
> on port 17230, notifying
> VideoMixer...
> -- VideoMixer (103)
> RTP-
> Listener for ConferenceID 8671000 started
> -- [CVM] Conference
> 8671000
> --> Session 1 / Peer 2 (pt 103)
> [Apr 7 16:51:19] NOTICE
> [11333]: rtp.c:
> 1256 ast_rtp_read: Unknown RTP codec 126 received from '192.168.23.240'
> [Apr 7 16:51:30] NOTICE[11333]: rtp.c:1256
> ast_rtp_read: Unknown RTP
> codec 126 received from '192.168.23.240'
> [Apr
> 7 16:51:40] NOTICE
> [11333]: rtp.c:1256 ast_rtp_read: Unknown RTP codec
> 126 received from
> '192.168.23.240'
> [Apr 7 16:51:50] NOTICE
> [11333]: rtp.
> c:1256
> ast_rtp_read: Unknown RTP codec 126 received from '192.168.23.240'
> [Apr
> 7 16:52:00] NOTICE[11333]: rtp.c:1256
> ast_rtp_read: Unknown RTP codec
> 126 received from '192.168.23.240'
> [Apr
> 7 16:52:10] NOTICE[11333]: rtp.
> c:1256 ast_rtp_read: Unknown RTP codec
> 126 received from
> '192.168.23.240'
> [Apr 7 16:52:20] NOTICE
> [11333]: rtp.
> c:1256
> ast_rtp_read: Unknown RTP codec 126 received from '192.168.23.240'
> [Apr
> 7 16:52:30] NOTICE[11333]: rtp.c:1256
> ast_rtp_read: Unknown RTP codec
> 126 received from '192.168.23.240'
> [Apr
> 7 16:52:41] NOTICE[11333]: rtp.
> c:1256 ast_rtp_read: Unknown RTP codec
> 126 received from
> '192.168.23.240'
> [Apr 7 16:52:51] NOTICE
> [11333]: rtp.
> c:1256
> ast_rtp_read: Unknown RTP codec 126 received from '192.168.23.240'
>
>
>
>
>
> ------------------------------------------------------------------------
> ----
> ------------------------------
>
>
>
>
>
> ##debug CONFIANCE VM IF (IN
> SIP.CONF):allow=h263
>
> maxcallbitrate=384
>
>
> [h263 @ 0xb7df8930]vbv
> buffer overflow
> [h263 @
> 0xb7df8930]vbv buffer overflow
> [h263 @
> 0xb7df8930]vbv buffer overflow
> [h263 @ 0xb7df8930]vbv buffer overflow
> [h263 @ 0xb7df8930]vbv buffer
> overflow
> [h263 @ 0xb7df8930]vbv buffer
> overflow
> [h263 @ 0xb7df8930]vbv
> buffer overflow
> [h263 @ 0xb7df8930]
> vbv
> buffer overflow
> [h263 @
> 0xb7df8930]vbv buffer overflow
> [h263 @
> 0xb7df8930]vbv buffer overflow
> [h263 @ 0xb7df8930]vbv buffer overflow
> [h263 @ 0xb7df8930]vbv buffer
> overflow
> [h263 @ 0xb7df8930]vbv buffer
> overflow
> [h263 @ 0xb7df8930]vbv
> buffer overflow
> [h263 @ 0xb7df8930]
> vbv
> buffer overflow
> [h263 @
> 0xb7df8930]vbv buffer overflow
> [h263 @
> 0xb7df8930]vbv buffer overflow
>
>
>
>
>
>
>
> ##debug di ASTERISK##
>
>
>
> *CLI>
> --
> Executing [8671000 at default:
> 1] Answer("SIP/101-081eb2f0", "") in
> new
> stack
> -- Executing
> [8671000 at default:2] MeetMe("SIP/101-081eb2f0",
> "8671000|B|") in new
> stack
> == Parsing '/etc/asterisk/xcon.conf': Found
> -- 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: 101,
> URI: sip:101 at 192.168.23.240:15709
> [Apr 7 16:57:42]
> WARNING[11390]:
> 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...
> -- [CVM]
> Conference 8671000 --> Session
> 1
> -- Started Video
> RTP Channel for user
> 1 on port 15486, notifying VideoMixer...
> --
> Format 524288 --> 34/H.263
> (confirmed)
> --
> Video Format: H.263
> --
> <SIP/101-081eb2f0> Playing 'conf-
> onlyperson'
> (language 'en')
> -- [CVM]
> User 1 (8671000) --> Session 1 /
> Peer 1
> -- Incoming H.263 (34) Video
> RTP Channel waiting on port 12024,
> notifying VideoMixer...
> -- ACK
> from
> XCON client received,
> requesting
> reinvite...
> -- Transmitting
> pending
> reinvite with BFCP
> information...
> -- Building
> SDP+BFCP/MSRP...
> --
> Actually
> sending reinvite with BFCP
> information...
> -- VideoMixer (34)
> RTP-
> Listener for ConferenceID
> 8671000 started
> -- [CVM] Conference
> 8671000 --> Session 1 / Peer 2 (pt
> 34)
> -- Parsing BFCP information
> in
> SIP OK's SDP: TCP/BFCP (bfcp port
> =
> 0, bound)...
> [Apr 7 16:57:43]
> NOTICE[11390]: rtp.c:1256
> ast_rtp_read:
> Unknown RTP codec 126 received from '192.168.23.240'
> [Apr 7 16:57:43]
> NOTICE[11390]: rtp.c:1256
> ast_rtp_read: Unknown RTP
> codec 126 received
> from
> '192.168.23.240'
> [Apr
> 7 16:57:43] NOTICE
> [11390]: rtp.c:1256
> ast_rtp_read: Unknown RTP codec
> 126 received from
> '192.168.23.240'
> [Apr
> 7 17:00:31] NOTICE
> [11377]:
> chan_sip.c:14761
> handle_request_subscribe:
> Received SIP
> subscribe for
> peer without
> mailbox: 101
>
>
> Can you help me
> to solve this probleme,
> please?
>
> I
> installed Confiance_videomixer end
> I'm using X-lite.
>
> Thanks
> Martina
>
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