[Asterisk-video] R: asterisk-video Digest, Vol 27, Issue 21
Marco Boldrin
marco.boldrin at spherait.it
Tue Jul 29 12:15:05 CDT 2008
Marco Boldrin
CIO & Solution Manager
Sphera IT Information Technology
www.spherait.it
Tel +390408326433 Int. 1001
Mob +393356996601
-----Original Message-----
From: asterisk-video-request at lists.digium.com
Date: Tue, 29 Jul 2008 12:00:05
To: <asterisk-video at lists.digium.com>
Subject: asterisk-video Digest, Vol 27, Issue 21
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Today's Topics:
1. Flash support (Sergio Garcia Murillo)
2. Re: How to find out if an incoming call has h324msupport
(Low Yu Siang)
3. got a bad video on minisip client (Mas Deva)
4. bad video quality on minisip client (Mas Deva)
5. Re: How to find out if an incoming call has h324msupport
(Klaus Darilion)
6. Re: How to find out if an incoming call has h324msupport
(Low Yu Siang)
7. Re: Flash support (Low Yu Siang)
8. Remotely disable video/audio stream in mobile phone (Low Yu Siang)
----------------------------------------------------------------------
Message: 1
Date: Mon, 28 Jul 2008 20:52:56 +0200
From: Sergio Garcia Murillo <sergio.garcia at fontventa.com>
Subject: [Asterisk-video] Flash support
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <488E1588.6040103 at fontventa.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi everyone,
As you may already know I'd been playing a bit with flash in the past.
I was thinking in finishing the development but first I would like to
pulse the interest on the following subjects:
-SWF playback.
This functionality would allow play a flash movie to an asterisk video
channel, mapping DMTF inputs to keys for user interaction.
-Flash video streaming
This functionality would allow to show an asterisk video channel on any
web page
-SIP-Flash gateway or asterisk flash channel
Best regards
Sergio
------------------------------
Message: 2
Date: Mon, 28 Jul 2008 19:59:37 -0700 (PDT)
From: Low Yu Siang <yusiang at yahoo.com>
Subject: Re: [Asterisk-video] How to find out if an incoming call has
h324msupport
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <685158.2350.qm at web65601.mail.ac4.yahoo.com>
Content-Type: text/plain; charset=utf-8
Any idea how to find out if I am using chan_ss7? It doesn't carry transfer capability parameter and the patch 0010217 is only for chan_zap....
--- On Mon, 28/7/08, Klaus Darilion <klaus.mailinglists at pernau.at> wrote:
> From: Klaus Darilion <klaus.mailinglists at pernau.at>
> Subject: Re: [Asterisk-video] How to find out if an incoming call has h324msupport
> To: "Development discussion of video media support in Asterisk" <asterisk-video at lists.digium.com>
> Date: Monday, 28 July, 2008, 3:20 PM
> Joost Kuif | Mobillion schrieb:
> > Hi Or,
> >
> > Use something like this:
> >
> > exten =>
> s,10,GotoIf($[${TRANSFERCAPABILITY}!=DIGITAL]?11:20)
>
> actually this checks only for digital calls, more accurate
> is to check
> the user information layer 1:
>
> CHANNEL(userinformationlayer1)!=38
>
> regards
> kalus
>
>
> >
> > Grtz,
> > Joost
> >
> >
> ------------------------------------------------------------------------
> > *Van:* asterisk-video-bounces at lists.digium.com
> > [mailto:asterisk-video-bounces at lists.digium.com]
> *Namens *FastAgi FastAgi
> > *Verzonden:* Sunday, July 27, 2008 1:00 PM
> > *Aan:* asterisk-video at lists.digium.com
> > *Onderwerp:* [Asterisk-video] How to find out if an
> incoming call has
> > h324msupport
> >
> > Hi,
> > I installed h324m support for asterisk, and it works
> great with phones
> > supporting 3g video calls.
> > I want to be able to distinguish phones that don't
> support 3g video
> > calls, send them to another context in the dialplan,
> and make that a
> > regular voice call.
> > Is there any way to do that?
> >
> > Thanks,
> > Or Agam
> >
> >
> >
> ------------------------------------------------------------------------
> >
> > _______________________________________________
> > --Bandwidth and Colocation Provided by
> http://www.api-digital.com--
> >
> > asterisk-video mailing list
> > To UNSUBSCRIBE or update options visit:
> >
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
> _______________________________________________
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> asterisk-video mailing list
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------------------------------
Message: 3
Date: Tue, 29 Jul 2008 11:01:10 +0700
From: "Mas Deva" <dvamust at gmail.com>
Subject: [Asterisk-video] got a bad video on minisip client
To: asterisk-video at lists.digium.com
Message-ID:
<5ab008890807282101h5646516fgdb86c5fb2b7a3671 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Now i'm in triying to build xcon in my lab.I think Asterisk server and video
mixer have already run normally.
But i got a problem when launch video. The result of the video that i
lounched with "videoswitch" command was so bad.(it was different from the
screnshoot on http://confiance.sourceforge.net/node/22).
here the display that i got from minisip log messages:
[h263 @ 0x1488650]Error at MB: 94
[h263 @ 0x1488650]concealing 81 DC, 81 AC, 81 MV errors
[h263 @ 0x1488650]Bad picture start code
[h263 @ 0x1488650]header damaged
[h263 @ 0x1488650]concealing 16 DC, 16 AC, 16 MV errors
[h263 @ 0x1488650]concealing 3 DC, 3 AC, 3 MV errors
[h263 @ 0x1488650]concealing 11 DC, 11 AC, 11 MV errors
[h263 @ 0x1488650]concealing 0 DC, 0 AC, 0 MV errors
[h263 @ 0x1488650]concealing 60 DC, 60 AC, 60 MV errors
[h263 @ 0x1488650]concealing 9 DC, 9 AC, 9 MV errors
[h263 @ 0x1488650]concealing 0 DC, 0 AC, 0 MV errors
[h263 @ 0x1488650]concealing 17 DC, 17 AC, 17 MV errors
[h263 @ 0x1488650]concealing 25 DC, 25 AC, 25 MV errors
[h263 @ 0x1488650]concealing 12 DC, 12 AC, 12 MV errors
[h263 @ 0x1488650]concealing 10 DC, 10 AC, 10 MV errors
[h263 @ 0x1488650]concealing 10 DC, 10 AC, 10 MV errors
[h263 @ 0x1488650]concealing 2 DC, 2 AC, 2 MV errors
[h263 @ 0x1488650]concealing 2 DC, 2 AC, 2 MV errors
[h263 @ 0x1488650]run overflow at 2x4 i:0
[h263 @ 0x1488650]Error at MB: 50
[h263 @ 0x1488650]run overflow at 10x2 i:0
[h263 @ 0x1488650]Error at MB: 34
[h263 @ 0x1488650]illegal ac vlc code at 4x3
[h263 @ 0x1488650]Error at MB: 40
[h263 @ 0x1488650]run overflow at 10x8 i:1
[h263 @ 0x1488650]Error at MB: 106
[h263 @ 0x1488650]concealing 99 DC, 99 AC, 99 MV errors
[h263 @ 0x1488650]run overflow at 1x2 i:1
[h263 @ 0x1488650]Error at MB: 25
[h263 @ 0x1488650]illegal ac vlc code at 6x4
[h263 @ 0x1488650]Error at MB: 54
[h263 @ 0x1488650]run overflow at 6x6 i:1
[h263 @ 0x1488650]Error at MB: 78
[h263 @ 0x1488650]concealing 91 DC, 91 AC, 91 MV errors
[h263 @ 0x1488650]concealing 16 DC, 16 AC, 16 MV errors
any suggestion Takashi?
regards
deva
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Message: 4
Date: Tue, 29 Jul 2008 11:07:41 +0700
From: "Mas Deva" <dvamust at gmail.com>
Subject: [Asterisk-video] bad video quality on minisip client
To: asterisk-video at lists.digium.com
Message-ID:
<5ab008890807282107n24f9d1eek7b9c4fec1df5aacf at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Dear all,
Now i'm in triying to build xcon in my lab.I think Asterisk server and video
mixer have already run normally.
But i got a problem when launch video. The result of the video that i
lounched with "videoswitch" command was so bad.(it was different from the
screnshoot on http://confiance.sourceforge.net/node/22).
here the display that i got from minisip log messages:
[h263 @ 0x1488650]Error at MB: 94
[h263 @ 0x1488650]concealing 81 DC, 81 AC, 81 MV errors
[h263 @ 0x1488650]Bad picture start code
[h263 @ 0x1488650]header damaged
[h263 @ 0x1488650]concealing 16 DC, 16 AC, 16 MV errors
[h263 @ 0x1488650]concealing 3 DC, 3 AC, 3 MV errors
[h263 @ 0x1488650]concealing 11 DC, 11 AC, 11 MV errors
[h263 @ 0x1488650]concealing 0 DC, 0 AC, 0 MV errors
[h263 @ 0x1488650]concealing 60 DC, 60 AC, 60 MV errors
[h263 @ 0x1488650]concealing 9 DC, 9 AC, 9 MV errors
[h263 @ 0x1488650]concealing 0 DC, 0 AC, 0 MV errors
[h263 @ 0x1488650]concealing 17 DC, 17 AC, 17 MV errors
[h263 @ 0x1488650]concealing 25 DC, 25 AC, 25 MV errors
[h263 @ 0x1488650]concealing 12 DC, 12 AC, 12 MV errors
[h263 @ 0x1488650]concealing 10 DC, 10 AC, 10 MV errors
[h263 @ 0x1488650]concealing 10 DC, 10 AC, 10 MV errors
[h263 @ 0x1488650]concealing 2 DC, 2 AC, 2 MV errors
[h263 @ 0x1488650]concealing 2 DC, 2 AC, 2 MV errors
[h263 @ 0x1488650]run overflow at 2x4 i:0
[h263 @ 0x1488650]Error at MB: 50
[h263 @ 0x1488650]run overflow at 10x2 i:0
[h263 @ 0x1488650]Error at MB: 34
[h263 @ 0x1488650]illegal ac vlc code at 4x3
[h263 @ 0x1488650]Error at MB: 40
[h263 @ 0x1488650]run overflow at 10x8 i:1
[h263 @ 0x1488650]Error at MB: 106
[h263 @ 0x1488650]concealing 99 DC, 99 AC, 99 MV errors
[h263 @ 0x1488650]run overflow at 1x2 i:1
[h263 @ 0x1488650]Error at MB: 25
[h263 @ 0x1488650]illegal ac vlc code at 6x4
[h263 @ 0x1488650]Error at MB: 54
[h263 @ 0x1488650]run overflow at 6x6 i:1
[h263 @ 0x1488650]Error at MB: 78
[h263 @ 0x1488650]concealing 91 DC, 91 AC, 91 MV errors
[h263 @ 0x1488650]concealing 16 DC, 16 AC, 16 MV errors
any suggestions?
regards
deva
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------------------------------
Message: 5
Date: Tue, 29 Jul 2008 10:16:36 +0200
From: Klaus Darilion <klaus.mailinglists at pernau.at>
Subject: Re: [Asterisk-video] How to find out if an incoming call has
h324msupport
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <488ED1E4.505 at pernau.at>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Low Yu Siang schrieb:
> Any idea how to find out if I am using chan_ss7? It doesn't carry
> transfer capability parameter and the patch 0010217 is only for
> chan_zap....
Hi! You are correct - reading the UL1 of incoming calls was not
implemented in chan_ss7. But it should be rather easy to do it yourself
- take a look at the patch for chan_zap and func_chan. The func_chan
part stays the same, you just have to port the chan_zap to chan_ss7 to
fill the channel structure with the UL1 information.
regards
klaus
------------------------------
Message: 6
Date: Tue, 29 Jul 2008 01:37:14 -0700 (PDT)
From: Low Yu Siang <yusiang at yahoo.com>
Subject: Re: [Asterisk-video] How to find out if an incoming call has
h324msupport
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <49668.44927.qm at web65614.mail.ac4.yahoo.com>
Content-Type: text/plain; charset=utf-8
For time being, I am using an easier patch(but less accurate).
In l4isup.c
if (inmsg->iam.trans_medium == 0x02) { /* 64kbit unrestricted data */
pvt->is_digital = 1;
+ pbx_builtin_setvar_helper(pvt->owner, "TRANSFERCAPABILITY", "DIGITAL");
+ } else {
+ pbx_builtin_setvar_helper(pvt->owner, "TRANSFERCAPABILITY", "SPEECH");
}
Will find some time to patch the UL1 later, thanks.
--- On Tue, 29/7/08, Klaus Darilion <klaus.mailinglists at pernau.at> wrote:
> From: Klaus Darilion <klaus.mailinglists at pernau.at>
> Subject: Re: [Asterisk-video] How to find out if an incoming call has h324msupport
> To: "Development discussion of video media support in Asterisk" <asterisk-video at lists.digium.com>
> Date: Tuesday, 29 July, 2008, 4:16 PM
> Low Yu Siang schrieb:
> > Any idea how to find out if I am using chan_ss7? It
> doesn't carry
> > transfer capability parameter and the patch 0010217 is
> only for
> > chan_zap....
>
> Hi! You are correct - reading the UL1 of incoming calls was
> not
> implemented in chan_ss7. But it should be rather easy to do
> it yourself
> - take a look at the patch for chan_zap and func_chan. The
> func_chan
> part stays the same, you just have to port the chan_zap to
> chan_ss7 to
> fill the channel structure with the UL1 information.
>
> regards
> klaus
>
> _______________________________________________
> --Bandwidth and Colocation Provided by
> http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
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------------------------------
Message: 7
Date: Tue, 29 Jul 2008 04:45:51 -0700 (PDT)
From: Low Yu Siang <yusiang at yahoo.com>
Subject: Re: [Asterisk-video] Flash support
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <943028.80960.qm at web65605.mail.ac4.yahoo.com>
Content-Type: text/plain; charset=utf-8
SIP-Flash gateway sounds very interesting! A few weeks ago, I was trying to get app_swf running, unfortunately ended with gnash showing some weird output.
--- On Tue, 29/7/08, Sergio Garcia Murillo <sergio.garcia at fontventa.com> wrote:
> From: Sergio Garcia Murillo <sergio.garcia at fontventa.com>
> Subject: [Asterisk-video] Flash support
> To: "Development discussion of video media support in Asterisk" <asterisk-video at lists.digium.com>
> Date: Tuesday, 29 July, 2008, 2:52 AM
> Hi everyone,
>
> As you may already know I'd been playing a bit with
> flash in the past.
>
> I was thinking in finishing the development but first I
> would like to
> pulse the interest on the following subjects:
>
> -SWF playback.
> This functionality would allow play a flash movie to an
> asterisk video
> channel, mapping DMTF inputs to keys for user interaction.
>
> -Flash video streaming
> This functionality would allow to show an asterisk video
> channel on any
> web page
>
> -SIP-Flash gateway or asterisk flash channel
>
> Best regards
> Sergio
>
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by
> http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
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------------------------------
Message: 8
Date: Tue, 29 Jul 2008 04:53:35 -0700 (PDT)
From: Low Yu Siang <yusiang at yahoo.com>
Subject: [Asterisk-video] Remotely disable video/audio stream in
mobile phone
To: Development discussion of video media support in Asterisk
<asterisk-video at lists.digium.com>
Message-ID: <387438.88377.qm at web65607.mail.ac4.yahoo.com>
Content-Type: text/plain; charset=utf-8
Hi all,
I understand that this question is not related to asterisk....Can anyone tell me if it is possible to remotely disable transmission of video/audio stream in the mobile phone via h324m(h245) protocol?
Regards,
Low Yu Siang
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