[Asterisk-video] R: Re: Asterisk Video-> debug Confiance videomixer
borja.sixto at i6net.com
borja.sixto at i6net.com
Wed Apr 9 10:29:07 CDT 2008
The App_conference have similar functions I think (You can play a file (wav)
from the CLI console, but the best way is to control that from a second legg).
I am going to make some tests next week (throw the h324m application...).
Regards,
Tech from i6net
Selon Sergio Garcia Murillo <sergio.garcia at fontventa.com>:
> 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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