23 Nov
2009
23 Nov
'09
3:30 p.m.
On Sun, Nov 22, 2009 at 09:38:42PM +0100, Esben Stien wrote:
torbenh <torbenh(a)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.)