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. -- :-- Hans Georg _______________________________________________ Linphone-users mailing list [email protected] https://lists.nongnu.org/mailman/listinfo/linphone-users
