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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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