gmane.comp.telephony.pbx.asterisk.video

Re: confbridge, h264 and switching

Dmitry Melekhov — 2012-04-10T03:52:06

09.04.2012 21:28, Matthew Jordan написал:
Yes, sure.
And this is what I'm asking for - do more :-)

Thank you!


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Re: confbridge, h264 and switching

Dmitry Melekhov — 2012-04-10T03:50:07

09.04.2012 19:15, Sergio Garcia Murillo написал:
Hello!


Well, I think so too :-)
Bad thing is I have no real   knowledge in this area, so I can't make 
patches ;-(



Thank you! We already tried it, looks good, but this solution needs 
really good server hardware, we don't have right now :-(


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Re: confbridge, h264 and switching

Sergio Garcia Murillo — 2012-04-09T22:28:10

Hi Matthew,

Checking one byte in first RTP packet of a frame shouldn't hurt the 
performance so badly.. ;)

Anyway, this is something that should be done in addition of requesting 
the update request in the INFO message.

Best regards
Sergio

El 09/04/2012 19:28, Matthew Jordan escribió:



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Re: confbridge, h264 and switching

Matthew Jordan — 2012-04-09T17:28:10

Sergio,

If you have to retrieve information from the RTP packet and make some
decision based on information within that packet - regardless of whether
or not you have to fully decode the h.264 information contained within
that packet - that's more then Asterisk currently does.

There's a trade-off between performance and capability here, and
performing logic on information in the media streams can quickly impact
performance, particularly when you can't define the system your
application will run on.

That being said, this is a scenario that - in our testing - most SIP
clients handle quite well with a request to update the picture.  If
that's the only issue Dmitry is running into, its most likely best
handled by the client, or determining why Asterisk is not sending
the necessary SIP INFO message.

If someone requires full media mixing, they're more then welcome to
look at other products.

Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: ht

Re: confbridge, h264 and switching

Sergio Garcia Murillo — 2012-04-09T15:15:08

Hi,

I/P frame information can be retrieved from the h263/h264 stream without 
decoding it.

Anyway, if you wan to try a full Multiconference server to integrate 
with Asterisk, you can try mine.. ;)

http://www.medooze.com/products/mcu.aspx

Best regards
Sergio

El 09/04/2012 16:34, Matthew Jordan escribió:



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Re: confbridge, h264 and switching

Matthew Jordan — 2012-04-09T14:34:43


----- Original Message -----

<snip>

Okay, now I understand what you're referring to.

Yes, if the client that has been made the new video source does not
transmit an I-Frame when they are made the video source, the receiving
clients will predict their frames on the previously received frames
from the previous video source.  That would lead to what you're
describing.

Since Asterisk does not interpret or decode the video streams it
receives from the clients, it does not have a mechanism itself to
manipulate the image.  It does, however, notify the client that is
the new video source via a SIP INFO message that it is now the source
of the video.  It is up to the client to know that it should then send
an I-Frame to provide a reference point for the other clients for the
new video images.

A few questions then:
1. What clients are you using?  Do they support RFC 5168 (XML Schema
for Media Control)?
2. If you get a SIP trace or a packet capture, do you see Asterisk
sending the SIP INFO messages when a video

Re: confbridge, h264 and switching

Dmitry Melekhov — 2012-04-09T03:55:53

06.04.2012 17:53, Matthew Jordan написал:
Hello!

I mean that without i-frame talking user appears from squares and other 
strange shape still showing part of previous talker.

Yes, I know, this will not make switching smoother.

Well, sort of.
Not  trancoding, but following to video stream and make switching on 
i-frame.
I guess this will not need much resources...
May be this is possible?

Thank you!


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Re: confbridge, h264 and switching

Matthew Jordan — 2012-04-06T13:53:28

Dmitry:

Can you explain what you mean by "it brokes during switching"?

ConfBridge has the capability to change the video source based on a
number of settings.  It can be configured to automatically set the
video source using talk detection, or based on the first/last marked
user that joined the conference.  Alternatively, you can use the DTMF
menu options to set the source of the video.

That being said, we do not have plans at this time to perform
transcoding on the video streams.  There's a number of reasons for this,
most notably that to do so is incredibly resource intensive.  Is that
what you were referring to?

Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

----- Original Message -----

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confbridge, h264 and switching

Dmitry Melekhov — 2012-04-06T04:31:31

Hello!

We tried to use confbridge for videoconferencing.
All looks good, but we use h264 and users complain about how video is 
switched, it brokes during switching, and this is expected behaviour,
just because asterisk do not analize video.
We also tried freeswitch and like asterisk ;-)
But I found that freeswitch tries (but really can't, I had to comment 
this out in sources) do video switching on i-frames.
Are there plans to implement such thing in asterisk's confbridge?

Thank you!


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Re: Help:app_rtsp cannot work in asterisk1.8

kingman chui — 2012-03-01T03:12:24

Hi,
I upload the wireshark capture packet log in below link

http://www.sendspace.com/file/t9lb6m

I already use the app_rtsp.c in sub dir 1.8 .
I capture the wireshark log in this email.

The IP 10.1.1.192 is the VLC rtsp server .
The IP 10.1.1.89 is the asterisk server .

The softphone also use 10.1.1.192 as I run it in the win 7 .

I can use VLC client and use rtsp to read the video from VLC rtsp serve without problem...

But when I use sofphone to estable the call , it is not work ...

The extensions.conf like below 

exten => 2002,1,Answer()
exten => 2002,n,rtsp(rtsp://10.1.1.192:5544/)
exten => 2002,n,WaitExten(5)
exten => 2002,n,Hangup()


Please advice what the problem is .

Thank


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Re: Help:app_rtsp cannot work in asterisk1.8

kingman chui — 2012-03-01T01:54:25

Hi,
  I upload the wireshark capture packet log in below link
 
http://www.sendspace.com/file/t9lb6m
 
 I already use the app_rtsp.c in sub dir 1.8 .
I capture the wireshark log in this email.

The IP 10.1.1.192 is the VLC rtsp server .
The IP 10.1.1.89 is the asterisk server .

The softphone also use 10.1.1.192 as I run it in the win 7 .

I can use VLC client and use rtsp to read the video from VLC rtsp serve without problem...

But when I use sofphone to estable the call , it is not work ...

The extensions.conf like below 

exten => 2002,1,Answer()
exten => 2002,n,rtsp(rtsp://10.1.1.192:5544/)
exten => 2002,n,WaitExten(5)
exten => 2002,n,Hangup()


Please advice what the problem is .

Thank



--- 2012年3月1日 星期四，kingman chui <chuikingman-/E1597aS9LTXPF5Rlphj1Q< at >public.gmane.org> 寫道﹕


寄件人: kingman chui <chuikingman-/E1597aS9LTXPF5Rlphj1Q< at >public.gmane.org>
主題: Re: [Asterisk-video] Help:app_rtsp cannot work in asterisk1.8
收件人: "Development discussion of video media supp

Re: Help:app_rtsp cannot work in asterisk1.8

kingman chui — 2012-03-01T01:37:57

Hi,
  The attached file for packet caputre log cannot post in the forum .
Please advice where I can put/upload the wireshark log file you want ???
 
I already use the app_rtsp.c under direcotry 1.8 ...
 
Thank
Regard/chui king man

--- 2012年2月22日 星期三，Sergio Garcia Murillo <sergio.garcia-f9lRAL0a+FOB+jHODAdFcQ< at >public.gmane.org> 寫道﹕


寄件人: Sergio Garcia Murillo <sergio.garcia< at >fontventa.com>
主題: Re: [Asterisk-video] Help:app_rtsp cannot work in asterisk1.8
收件人: asterisk-video-Ra3b/QYEcJ2raY3XHr1+6Q< at >public.gmane.org
日期: 2012年2月22日,星期三,上午7:00



Hi,

Are you using the app_rtsp under the 1.8 subdirectory? Also, could you get an ethreal cpature of the RSTP connection?

Best regards
Sergio

El 20/02/2012 14:49, kingman chui escribió: 









Hi,
  I install app_rtsp in 1.8.10.0-rc2  .
I cannot connect to rtsp VLC streaming server .
the rtsp VLC server is working find with VLC client and can play video in H263 format.
no audio .I disable audio .
 
I pas

Re: Asterisk crash when mp4play run 3gp file

2012-02-29T09:21:12

Hi,

This problem hapenned on old mp4v2 lib versions (in fact i did the patch to solve it).

Please, get and compile latest version of the mp4v2 repository.

Enviado desde mi HTC

----- Reply message -----
De: "Aris Striglis" <astriglis< at >gmail.com>
Fecha: mar., feb. 28, 2012 21:13
Asunto: [Asterisk-video] Asterisk crash when mp4play run 3gp file
Para: "'Development discussion of video media support in Asterisk'" <asterisk-video< at >lists.digium.com>

Hello,

 

I’m using misdn from my BRI,

I have succeeded in making a video_loopback and echo test,  3g phone ->asterisk

using  dialplan below:

 

exten =>  ,1,GotoIf($[${TRANSFERCAPABILITY}=DIGITAL]?10:20)

exten =>  ,10,h324m_gw(310< at >tosip)

exten =>  ,20,Dial(SIP/202)

exten =>  ,n,Playback(noaudiocall)

exten =>  ,n,Hangup()

 

 

[tosip]

;exten => 310,1,h324m_gw_answer()

;exten => 310,n,Echo()                                                  ß--working!!

;exten => 310,n,video_loopback()                           ß--working!!

;exten => 310,n,Dial(SIP/x-

Asterisk crash when mp4play run 3gp file

Aris Striglis — 2012-02-28T20:13:50

Hello,

 

I’m using misdn from my BRI,

I have succeeded in making a video_loopback and echo test,  3g phone ->asterisk

using  dialplan below:

 

exten =>  ,1,GotoIf($[${TRANSFERCAPABILITY}=DIGITAL]?10:20)

exten =>  ,10,h324m_gw(310< at >tosip)

exten =>  ,20,Dial(SIP/202)

exten =>  ,n,Playback(noaudiocall)

exten =>  ,n,Hangup()

 

 

[tosip]

;exten => 310,1,h324m_gw_answer()

;exten => 310,n,Echo()                                                  ß--working!!

;exten => 310,n,video_loopback()                           ß--working!!

;exten => 310,n,Dial(SIP/x-lite)                                  ß--incoming video call // working!!

;exten => 310,n,Hangup()

 

 

Now, i am trying to play a video using mp4play to a 3g phone calling my asterisk using the following dialplan:

 

exten => 310,1,h324m_gw_answer()

exten => 310,n,mp4play(/root/mire-tv.3gp)

 

But as soon as call comes in i get disconnected from asterisk cli and 3g phone still waiting the video.

The output from CLI, shows that app_mp4 c

Re: Quest for working MCU software

2012-02-27T17:44:03

Hi Ray,

You have mismatch of version in tne mcu media mixer and on the mcuWeb. Plase, make sure u are using latest versions.

Br
Sergio

Enviado desde mi HTC

----- Reply message -----
De: "ray klassen" <julius_ahenobarbus< at >yahoo.co.uk>
Fecha: lun., feb. 27, 2012 17:07
Asunto: [Asterisk-video] Quest for working MCU software
Para: "Development discussion of video media support in Asterisk" <asterisk-video< at >lists.digium.com>

[#|2012-02-27T08:05:03.424-0800|SEVERE|sun-glassfish-comms-server2.0|global|_ThreadID=19;_ThreadName=SipContainer-serversWorkerThread-5060-7;_RequestID=e47b8bc5-7167-47c5-b34d-9346816342af;|The log message is null.
org.apache.xmlrpc.XmlRpcException: Format string requests exactly 3 items from array, but array has 4 items.  (A '*' at the end would avoid this failure)






----- Original Message -----
From: ray klassen <julius_ahenobarbus< at >yahoo.co.uk>
To: Development discussion of video media support in Asterisk <asterisk-video< at >lists.digium.com>
Cc: 
Sent: Friday, 24 February 2012, 16:46
S

Re: Quest for working MCU software

ray klassen — 2012-02-27T16:07:59

[#|2012-02-27T08:05:03.424-0800|SEVERE|sun-glassfish-comms-server2.0|global|_ThreadID=19;_ThreadName=SipContainer-serversWorkerThread-5060-7;_RequestID=e47b8bc5-7167-47c5-b34d-9346816342af;|The log message is null.
org.apache.xmlrpc.XmlRpcException: Format string requests exactly 3 items from array, but array has 4 items.  (A '*' at the end would avoid this failure)






----- Original Message -----
From: ray klassen <julius_ahenobarbus-/E1597aS9LT10XsdtD+oqA< at >public.gmane.org>
To: Development discussion of video media support in Asterisk <asterisk-video-Ra3b/QYEcJ2raY3XHr1+6Q< at >public.gmane.org>
Cc: 
Sent: Friday, 24 February 2012, 16:46
Subject: Re: [Asterisk-video] Quest for working MCU software

Moving on
I found where to set the sip listener in sailfin, and set it for the right ip. UDP connection issue solved. 

Now I'm getting circuits are busy still but the error report is now

   -- Got SIP response 500 "Server Internal Error" back from Random-IP-address:5060

(Thanks for all your help. )

 



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Re: Quest for working MCU software

Sergio Garcia Murillo — 2012-02-25T22:53:48

What i the trace at Sailfin server.log?

Best regards
Sergio
El 25/02/2012 1:46, ray klassen escribió:



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Re: Quest for working MCU software

ray klassen — 2012-02-25T00:46:14

Moving on
I found where to set the sip listener in sailfin, and set it for the right ip. UDP connection issue solved. 

Now I'm getting circuits are busy still but the error report is now

   -- Got SIP response 500 "Server Internal Error" back from Random-IP-address:5060

(Thanks for all your help. )

 



----- Original Message -----
From: ray klassen <julius_ahenobarbus-/E1597aS9LT10XsdtD+oqA< at >public.gmane.org>
To: Development discussion of video media support in Asterisk <asterisk-video-Ra3b/QYEcJ2raY3XHr1+6Q< at >public.gmane.org>
Cc: 
Sent: Wednesday, 22 February 2012, 8:14
Subject: Re: [Asterisk-video] Quest for working MCU software

Yes it's sending it to the right port. On the mcuWeb end netstat -aunp shows no port udp port 5060 even open even though it reports that it is listening on those ports when you restart the server.



----- Original Message -----
From: Sergio Garcia Murillo <sergio.garcia-f9lRAL0a+FOB+jHODAdFcQ< at >public.gmane.org>
To: asterisk-video-Ra3b/QYEcJ2raY3XHr1+6Q< at >public.gmane.org
Cc:

Re: Quest for working MCU software

Sergio Garcia Murillo — 2012-02-22T16:20:25

Hi,

Could paste the output of the netstat here? What port is configured on 
the Sailfin admin page for both sip listener and sip contained? Are you 
running Asterisk on the same server?

Best regards
Sergio

El 22/02/2012 17:14, ray klassen escribió:



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Re: Quest for working MCU software

ray klassen — 2012-02-22T16:14:47

Yes it's sending it to the right port. On the mcuWeb end netstat -aunp shows no port udp port 5060 even open even though it reports that it is listening on those ports when you restart the server.

----- Original Message -----
From: Sergio Garcia Murillo <sergio.garcia-f9lRAL0a+FOB+jHODAdFcQ< at >public.gmane.org>
To: asterisk-video-Ra3b/QYEcJ2raY3XHr1+6Q< at >public.gmane.org
Cc: 
Sent: Tuesday, 21 February 2012, 14:58
Subject: Re: [Asterisk-video] Quest for working MCU software

Hi,

What is the Asterisk error? Could check if it is sending the INVITE to 
the sailfin IP:port?

Best regards
Sergio

El 21/02/2012 20:26, ray klassen escribió:

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Re: Help:app_rtsp cannot work in asterisk1.8

Sergio Garcia Murillo — 2012-02-21T23:00:14

Hi,

Are you using the app_rtsp under the 1.8 subdirectory? Also, could you 
get an ethreal cpature of the RSTP connection?

Best regards
Sergio

El 20/02/2012 14:49, kingman chui escribió:

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