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Re: [Linphone-users] RTP relaying


From: Ghislain MARY
Subject: Re: [Linphone-users] RTP relaying
Date: Fri, 02 May 2014 15:07:24 +0200
User-agent: Mozilla/5.0 (X11; Linux x86_64; rv:24.0) Gecko/20100101 Icedove/24.4.0

Hi,

By default, Linphone uses the SIP server as a media relay as this is the only way to be sure that the media goes through in case the clients do not have a public IP (are behind NAT). However, you can enable the ICE feature of Linphone to prevent using the media relay. The purpose of ICE is to find the most direct path for the media by sending some STUN packets. So you need to configure a STUN server (such as stun.linphone.org) in your Linphone configuration on both clients and enable ICE on both clients also. The media relay will still be used for a little time at the beginning of the call until ICE has found a better path for the media (if it finds one).

Cheers,
Ghislain

On 02/05/2014 11:30, Hans Georg Schaathun wrote:
Dear all,

could someone help shed a little bit of light on the use of the
SIP server as a relay for RTP (video/audio) traffic?

Using Linphone and sip.linphone.org, I observe that all traffic
is routed via the SIP server, even when both clients have public IP
addresses (no NAT) and should be able to route directly (once the
session is initiatied).

Is it the server or the clients who decide whether to use RTP
relaying?  I was not able to find any configuration option for it
in the android client, but I am not sure I have the right jargon
to find it.  Is it possible to change the setting?

If I understand it correctly, the RTP relaying is not covered by
SIP.  Is there an open standard to explain what is going on?
In particular, will the the server be able to route any video/audio
stream oblivious to the codec in use?

Thank you very much for your help.




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