Hi,
On 10/13/14 11:26 AM, Henrique Rosa wrote:
Hello,
I've been working on WebRTC support for Mobicents Media Server (MMS).
I already succeeded at interop between MMS and Firefox/Chrome.
Unfortunately, while testing conference calls I noticed that there is a
tendency for RTT times and Delay to increase as the call goes on (on both
browsers).
On Chrome, the delay ranges from 3 to 10 seconds for a conference call between
two participants with duration of 3-4 minutes.
On Firefox the results are much better: the delay ranges from 1 to 3 seconds in
the same scenario.
Note that while testing with regular SIP clients (jitsi, linphone, xlite) there
is no delay whatsoever, so the issue is related with WebRTC calls.
My question is: what can possibly cause such delay?
The one obvious difference between WebRTC and "regular SIP clients" is
the additional media encryption. But should not take seconds, at least
if you are using modern desktop PC.
The other question is how much delay FF and Chrome add between capturing
audio and actually sending them out on the wire. And on the other end
how delay there is between receiving audio packets and rendering to your
audio device.
Best regards
Nils Ohlmeier
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