[Asterisk-video] call handshake fails

[Emmanuel BUU]

That 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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