[Asterisk-video] call handshake fails

Zelalem Sintayehu zelalems at hotmail.com
Tue Dec 2 03:21:41 CST 2008


Hi Klaus and Carlo, thank you for your response. There is a little development. X-Lite seemed to save the video but couldn't play it back. You know I had to manually clicked the start button on the video pannel to start sending the video. Then, I have looked at the file using mp4info and got the following info:
  Track    Type    Info
1    audio    G.711 uLaw, 5.940 secs, 64 kbps, 8000 Hz
2    hint    Payload PCMU for track 1
3    video    H.263, 5.844 secs, 835 kbps, 176x144 @ 26.694045 fps
4    hint    Payload H263-1998 for track 3

I couldn't play it back, though. But I have seen the file using Ubuntu's movie player and it is a very scrambled video. I am using logitech quickcam sphere mp webcam. After, I read Carlo's e-mail, then I came back to ubuntu and tried linphone. Again, I found out that I didn't set the minimum upload and download bandwidth. So, I set 135 kbit/s for video (I am using h263-1998 codec). Then it showed wierd characterisk (like terminating automatically and not starting again). I got the following error messge from Asterisk when I tried to record the video:
    -- Executing [7504 at jain-sip:1] Answer("SIP/7503-b7b11730", "") in new stack
    -- Executing [7504 at jain-sip:2] mp4save("SIP/7503-b7b11730", "/home/zelalem/videos/save.mp4") in new stack
[Dec  2 18:47:54] WARNING[15692]: channel.c:2809 set_format: Unable to find a codec translation path from h263p to unknown
[Dec  2 18:47:54] WARNING[15692]: app_mp4.c:804 mp4_save: mp4_save: Unable to set read format to ULAW|ALAW|AMRNB!

And the phone terminated. I hope the above gives you an idea as to what my problem is. Once again, thank you. I have been doing this the past two weeks.

Cheers,

- Zelalem S. 
Grahamstown, SA



> Date: Fri, 28 Nov 2008 10:40:55 +0100
> From: klaus.mailinglists at pernau.at
> To: asterisk-video at lists.digium.com
> Subject: Re: [Asterisk-video] call handshake fails
> 
> Hi!
> 
> Thanks for the patch. Now it also works with a Sony Ericsson V800.
> 
> regards
> Klaus
> 
> 
> 
> Dan Julius schrieb:
> > Hi, Sergio,
> > 
> > I'm following up on the problem I've been having that some (30% - 60%, 
> > not sure what it depends on) calls are not connected successfully when 
> > dialing from Samsung 3G phone to SIP client.
> > 
> > Turns out that increasing the retransmit delay from 20 to 2000 in 
> > H324CCSRLayer::GetNextPdu seems to have resolved the problem.
> > 
> > Are these units Milliseconds?
> > What do you think should be a reasonable timeout?
> > 
> > Dan
> > 
> > 
> > On Thu, May 15, 2008 at 4:42 PM, Dan Julius <dan.julius at gmail.com 
> > <mailto:dan.julius at gmail.com>> wrote:
> > 
> >     Hi, Sergio,
> > 
> >     I actually sent these to the list a while ago, but they bounced.
> >     How do we deal with private attachments while still keeping the
> >     discussion public?
> > 
> >     Thanks for looking into this.
> > 
> >     Dan
> > 
> >     ---------- Forwarded message ----------
> >     From: *Dan Julius* <dan.julius at gmail.com <mailto:dan.julius at gmail.com>>
> >     Date: Fri, May 9, 2008 at 2:07 PM
> >     Subject: Re: [Asterisk-video] call handshake fails
> >     To: Development discussion of video media support in Asterisk
> >     <asterisk-video at lists.digium.com
> >     <mailto:asterisk-video at lists.digium.com>>
> > 
> > 
> >     Hi,
> > 
> >     Attached are logs for a call that failed. After answering the call
> >     on the mobile device, X-Lite continues to ring and nothing happens.
> >     As for video in working calls - the problem is with video from H324M
> >     to SIP. Any ideas how to debug this?
> > 
> >     Can you provide a sample for using app_transcoder?
> > 
> >     Thanks,
> >     Dan
> > 
> > 
> > 
> >     On Fri, May 9, 2008 at 1:27 PM, Sergio Garcia Murillo
> >     <sergio.garcia at fontventa.com <mailto:sergio.garcia at fontventa.com>>
> >     wrote:
> > 
> >         Could you send me a file with the h245 and h223 logs? (enable
> >         them by h324m debug level 4)
> > 
> >         The most probable cause is that you isdn provider is doing echo
> >         cancelation on the line, it usually causes random problems like
> >         this.
> > 
> >         The problem with video from SIP->H324M is that it has to be h263
> >         QCIF at maximun 52 kbs, if your videophone is not able to set
> >         this up, you'll need to use the app_transcoder module.
> > 
> >         Best regards
> >         Sergio
> > 
> >         ----- Original Message -----
> >         From: Dan Julius [mailto:dan.julius at gmail.com
> >         <mailto:dan.julius at gmail.com>]
> >         To: asterisk-video at lists.digium.com
> >         <mailto:asterisk-video at lists.digium.com>
> >         Sent: Fri, 9 May 2008 12:25:17 +0300
> >         Subject: Re: [Asterisk-video] call handshake fails
> > 
> >         Further info:
> > 
> >         - In the failed calls, the mobile phone never sends a
> >         masterSlaveDetermination packet (according to the h223 logs)
> >         - Asterisk sends the terminalCapabilitiesSet,
> >         masterSlaveDetermination and
> >         then continues to send OpenLogicalChannels.
> > 
> >         Is it OK to send OpenLogicalChannel before receiving a
> >         masterSlaveDetermination?
> > 
> >         Thanks,
> >         Dan
> > 
> >         On Fri, May 9, 2008 at 2:25 AM, Dan Julius <dan.julius at gmail.com
> >         <mailto:dan.julius at gmail.com>> wrote:
> > 
> >          > Hi, Everybody,
> >          >
> >          > I'm new to this project, so  I apologize if  my questions
> >         might have
> >          > already been answered elsewhere.
> >          > I am using a X-Lite, Asterisk 1.4.19, a Digium TE122 card,
> >         and a Samsung
> >          > Z720 phone.
> >          >
> >          > So far I have been able to make SIP-h234m calls (originating
> >         at either
> >          > side) with only partial success.
> >          > - I only get video in one direction, from SIP to H324M. I've
> >         read the posts
> >          > stating that SIP->H324m is actually more problematic, so I'm
> >         quite puzzled
> >          > about this.
> >          > - About 33% of the calls fail to negotiate a video
> >         connection. After
> >          > answering the call, nothing happens until I disconnect.
> >          > The out-bound h223 log of a failed call is below. Does this
> >         log indicate
> >          > that Asterisk is sending terminalCapabilitySet multiple times
> >         until it is
> >          > acknowledged?
> >          >
> >          > 1   0.000000      1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> >         <http://2.2.2.2>      H.245 terminalCapabilitySet
> >          > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet
> >          > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet
> >          > terminalCapabilitySet terminalCapabilitySet terminalCapabilitySet
> >          > masterSlaveDetermination masterSlaveDetermination
> >         masterSlaveDetermination
> >          > masterSlaveDetermination masterSlaveDetermination
> >         masterSlaveDetermination
> >          > masterSlaveDetermination masterSlaveDetermination
> >         masterSlaveDetermination
> >          > masterSlaveDetermination masterSlaveDetermination
> >         masterSlaveDetermination
> >          > masterSlaveDetermination masterSlaveDetermination
> >         masterSlaveDetermination
> >          >   2   0.000001      1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> >         <http://2.2.2.2>      H.245 openLogicalChannel
> >          > (generic) openLogicalChannel (generic) openLogicalChannel
> >         (generic)
> >          > openLogicalChannel (generic) openLogicalChannel (generic)
> >         openLogicalChannel
> >          > (generic) openLogicalChannel (generic) openLogicalChannel
> >         (generic)
> >          > openLogicalChannel (generic) openLogicalChannel (generic)
> >         openLogicalChannel
> >          > (generic) openLogicalChannel (generic) openLogicalChannel
> >         (generic)
> >          > openLogicalChannel (h263VideoCapability) openLogicalChannel
> >          > (h263VideoCapability) openLogicalChannel (h263VideoCapability)
> >          > openLogicalChannel (h263VideoCapability) openLogicalChannel
> >          > (h263VideoCapability) openLogicalChannel (h263VideoCapability)
> >          > openLogicalChannel (h263VideoCapability) openLogicalChannel
> >          > (h263VideoCapability) openLogicalChannel (h263VideoCapability)
> >          > openLogicalChannel (h263VideoCapability) openLogicalChannel
> >          > (h263VideoCapability) openLogicalChannel (h263VideoCapability)
> >          > openLogicalChannel (h263VideoCapability) openLogicalChannel
> >          > (h263VideoCapability) multiplexEntrySend multiplexEntrySend
> >          > multiplexEntrySend multiplexEntrySend multiplexEntrySend
> >         multiplexEntrySend
> >          > multiplexEntrySend multiplexEntrySend multiplexEntrySend
> >         multiplexEntrySend
> >          > multiplexEntrySend multiplexEntrySend
> >          >   3   0.000002      1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> >         <http://2.2.2.2>      H.245 multiplexEntrySend
> >          > multiplexEntrySend multiplexEntrySend multiplexEntrySend
> >          > terminalCapabilitySetAck terminalCapabilitySetAck
> >         terminalCapabilitySetAck
> >          > terminalCapabilitySetAck terminalCapabilitySetAck
> >         terminalCapabilitySetAck
> >          > terminalCapabilitySetAck terminalCapabilitySetAck
> >         terminalCapabilitySetAck
> >          > terminalCapabilitySetAck terminalCapabilitySetAck
> >         terminalCapabilitySetAck
> >          > terminalCapabilitySetAck terminalCapabilitySetAck
> >         terminalCapabilitySetAck
> >          > terminalCapabilitySetAck
> >          >   4   0.000003      1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> >         <http://2.2.2.2>      H223
> >          >   5   0.000004      1.1.1.1 <http://1.1.1.1> -> 2.2.2.2
> >         <http://2.2.2.2>      H223
> >          >
> >          > Any pointers on how to debug this would be much appreciated.
> >          >
> >          > Thanks,
> >          > Dan
> >          >
> >          > PS - This is really great work and I'm very impressed with
> >         the project and
> >          > hope that I will be able to contribute as well.
> >          >
> >          >
> >          >
> >          >
> >          >
> >          >
> >          >
> > 
> > 
> >         _______________________________________________
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> > 
> >         asterisk-video mailing list
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> >           http://lists.digium.com/mailman/listinfo/asterisk-video
> > 
> > 
> > 
> > 
> > 
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> > 
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