[LAU] Community Conference Room

[Ken Restivo]

On Sun, Nov 22, 2009 at 09:38:42PM +0100, Esben Stien wrote:
> torbenh <torbenh at gmx.de> 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.)
>