[Asterisk-video] call handshake fails

[Zelalem Sintayehu]

Hi Emanuel, it makes sense. Because I get this error message (from Asterisk) immediately after the connection is established. But I can't get where I can enable/disable this in my X-Lite. X-Lite has only settings related to echo and noise cancellation.
 
Thanks again for your support.
- Zelalem S. Grahamstown, SA



Date: Wed, 3 Dec 2008 18:28:42 +0100From: emmanuel.buu at ives.frTo: asterisk-video at lists.digium.comSubject: Re: [Asterisk-video] call handshake failsThat is most probably silence packets that are sent by yout X-Lite clients and not announced in the SDP.No influence so far on the operation.Zelalem Sintayehu a écrit : 
Hi Klaus and Carlo, it is now working in Windows (X-Lite). I had to remove the 263+ video coved. I have now only the h263 codec. I think 263+ (i.e, h263-1998) may be sending bad code to asterisk (or it may be the client). In fact, asterisk terminates when i try to playback the demo file (which is encoded with h263-2000). You know, I have been trying different things since Monday, but after I remove the codec, it started working properly. But, I have still the following error message (uknown rtp codec 126 ...) in Asterisk.     -- Executing [2000 at jain-sip:1] Answer("SIP/7503-08204e98", "") in new stack    -- Executing [2000 at jain-sip:2] mp4play("SIP/7503-08204e98", "/home/zelalem/videos/save.mp4") in new stack[Dec  4 01:16:21] NOTICE[21538]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '146.231.122.97'[Dec  4 01:16:21] NOTICE[21538]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '146.231.122.97'[Dec  4 01:16:21] NOTICE[21538]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '146.231.122.97'[Dec  4 01:16:31] NOTICE[21538]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '146.231.122.97'    -- Executing [2000 at jain-sip:3] Hangup("SIP/7503-08204e98", "") in new stack  == Spawn extension (jain-sip, 2000, 3) exited non-zero on 'SIP/7503-08204e98'
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