[Asterisk-video] video prompts via Asterisk

[Sergio García Murillo]

Yes, sending a ceroed rtp header should do the trick in almost any device. I didn't found any issue when i used it. 
 
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From: asterisk-video-bounces at lists.digium.com [mailto:asterisk-video-bounces at lists.digium.com] On Behalf Of Ramtin Amin
Sent: martes, 28 de noviembre de 2006 22:01
To: Development discussion of video media support in Asterisk
Subject: RE: [Asterisk-video] video prompts via Asterisk




Stun packets are regognized as being RTP v0 so if the two first bits are 00xxxxxx then it's STUN/ICE... if you have 10xxxxxx it's RTP v2... so quiet easy to discard




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	Date: Tue, 28 Nov 2006 12:55:56 -0800
	From: duane at counterpath.com
	To: asterisk-video at lists.digium.com
	Subject: Re: [Asterisk-video] video prompts via Asterisk
	
	Even if you don't support parsing the STUN packets, you should make sure you will discard them anyways (since a STUN packet will fail as a RTP packet if you do a proper header check).
	
	For video doing keep alives is a little trickier than audio -- in audio you can encode silence and send that as data, knowing the remote party won't hear it.  The equivalent in video doesn't really exist.  So, STUN packets are great if you can use them, otherwise I've seen some implementations send a bogus RTP packet using an unused payload type to keep the pinhole open. 
	
	Duane
	
	
	On 11/28/06, Olle E Johansson <oej at edvina.net> wrote: 


		28 nov 2006 kl. 15.42 skrev Bruce Bauman:
		
		> If a videophone is behind a NAT, the firewall will often timeout the
		> bindings after a small amount of time and Asterisk will be unable to
		> send video to the phone. 
		>
		> Has anyone encountered/solved this problem? How do you keep the RTP
		> stream open during periods of inactivity?
		
		There's something called RTP keepalives that some devices use.
		You need to send at least every 29th sec... 
		
		What it really is, is something for others to discuss. With newer
		devices
		you could send STUN requests on the RTP port - you certainly have to
		be ready to receive them (the new SIP outbound draft).
		
		/O
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