On Sun, Nov 22, 2009 at 09:38:42PM +0100, Esben Stien wrote: > torbenh <torbenh@xxxxxx> writes: > > > does sip support measuring the latency of the connection ? I use Twinkle for my SIP calls, and it does measure the latency of the connection, as well as packet loss, which is significant for me most of the time. > > SIP doesn't really deal with that. SIP is a session initiation protocol, > a session of anything really and in VoIP, it's basically two RTP > channels and a SIP control channel. > > The Real-Time Control Protocol (RTCP), a companion protocol to RTP, is > used by applications to monitor the delivery of RTP streams. Media > packets are transmitted between endpoints during a session according to > RTP while additional performance information governing the communication > link (e.g., key statistics about the media packets being sent and > received by each endpoint such as jitter, packet loss, round-trip time, > etc.) > _______________________________________________ Linux-audio-user mailing list Linux-audio-user@xxxxxxxxxxxxxxxxxxxx http://lists.linuxaudio.org/mailman/listinfo/linux-audio-user
