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[Asterisk-video] R: Re: Asterisk Video->debug Confiance v

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sergio.garcia at fontvent
Guest

Posted: Wed Apr 09, 2008 11:19 am Post subject: [Asterisk-video] R: Re: Asterisk Video->debug Confiance v

Quote:

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@gmail.
com
Data: 8-apr-2008 3.23
PM
A: "Development discussion of video media
support in Asterisk"
<asterisk-video@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@lists.
digium.com
[mailto:
asterisk-video-bounces@lists.digium.com] En nombre
de marti2001@tin.it
Enviado el: martes, 08 de abril de 2008 15:05
Para:
asterisk-
video@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@default:1] Answer("SIP/101-
081cba58", "") in new
stack
-- Executing [8671000@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@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)
--

Quote:

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@default:
1] Answer("SIP/101-081eb2f0", "") in
new
stack
-- Executing
[8671000@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@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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sergio.garcia at fontvent
Guest

Posted: Wed Apr 09, 2008 2:21 pm Post subject: [Asterisk-video] R: Re: Asterisk Video->debug Confiance v

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@i6net.com [mailto:borja.sixto@i6net.com]
To: asterisk-video@lists.digium.com,sergio.garcia@fontventa.com
Cc: asterisk-video@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@fontventa.com>:

Quote:

You could try my multiconference solution (mcuWeb+MediaMixer)

http://sip.fontventa.com/content/view/32/63/

Best regards
Sergio

----- Original Message -----
From: marti2001@tin.it [mailto:marti2001@tin.it]
To: asterisk-video@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@gmail.com
Data: 8-apr-2008 11.48 PM
A: "Development
discussion of video media support in Asterisk"<asterisk-video@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@lists.digium.com
[mailto:
asterisk-video-bounces@lists.digium.com] En nombre de
marti2001@tin.it
Enviado el: martes, 08 de abril de 2008 18:56
Para: asterisk-
video@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@gmail.
com
Data: 8-apr-2008 3.23
PM
A: "Development discussion of video media
support in Asterisk"
<asterisk-video@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@lists.
digium.com
[mailto:
asterisk-video-bounces@lists.digium.com] En nombre
de marti2001@tin.it
Enviado el: martes, 08 de abril de 2008 15:05
Para:
asterisk-
video@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@default:1] Answer("SIP/101-
081cba58", "") in new
stack
-- Executing [8671000@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@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@default:
1] Answer("SIP/101-081eb2f0", "") in
new
stack
-- Executing
[8671000@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@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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