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<DIV dir=ltr align=left><SPAN class=020195607-29112006>Yes, sending a ceroed rtp
header should do the trick in almost any device. I didn't found any issue when i
used it. </SPAN></DIV>
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<B>From:</B> asterisk-video-bounces@lists.digium.com
[mailto:asterisk-video-bounces@lists.digium.com] <B>On Behalf Of </B>Ramtin
Amin<BR><B>Sent:</B> martes, 28 de noviembre de 2006 22:01<BR><B>To:</B>
Development discussion of video media support in Asterisk<BR><B>Subject:</B> RE:
[Asterisk-video] video prompts via Asterisk<BR><BR></DIV>
<DIV></DIV><BR><BR>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<BR><BR><BR>
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Date: Tue, 28 Nov 2006 12:55:56 -0800<BR>From: duane@counterpath.com<BR>To:
asterisk-video@lists.digium.com<BR>Subject: Re: [Asterisk-video] video prompts
via Asterisk<BR><BR>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).<BR><BR>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.
<BR><BR>Duane<BR><BR>
<DIV><SPAN class=EC_gmail_quote>On 11/28/06, <B class=EC_gmail_sendername>Olle
E Johansson</B> <<A href="mailto:oej@edvina.net">oej@edvina.net</A>>
wrote:</SPAN>
<BLOCKQUOTE class=EC_gmail_quote style="PADDING-LEFT: 1ex"><BR>28 nov 2006
kl. 15.42 skrev Bruce Bauman:<BR><BR>> If a videophone is behind a NAT,
the firewall will often timeout the<BR>> bindings after a small amount of
time and Asterisk will be unable to<BR>> send video to the phone.
<BR>><BR>> Has anyone encountered/solved this problem? How do you keep
the RTP<BR>> stream open during periods of inactivity?<BR><BR>There's
something called RTP keepalives that some devices use.<BR>You need to send
at least every 29th sec... <BR><BR>What it really is, is something for
others to discuss. With newer<BR>devices<BR>you could send STUN requests on
the RTP port - you certainly have to<BR>be ready to receive them (the new
SIP outbound
draft).<BR><BR>/O<BR>_______________________________________________<BR>--Bandwidth
and Colocation provided by <A href="http://easynews.com/"
target=_blank>Easynews.com</A> --<BR><BR>asterisk-video mailing list<BR>To
UNSUBSCRIBE or update options visit: <BR> <A
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target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-video</A><BR></BLOCKQUOTE></DIV><BR></BLOCKQUOTE><BR>
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