[OpenSIPS-Users] Sound jitter

Vlad Paiu vladpaiu at opensips.org
Thu Sep 19 18:09:33 CEST 2013


RTP proxy is just a proxy - it just blindly relays RTP packages - and 
doesn't have transcoding capabilities and thus is very light weight and 
should not introduce and jitter on it's own.

You might have a high speed connection, but it also depends on the 
client's connection, and also on the quality of the link/path between 
you and the client.

Best Regards,

Vlad Paiu
OpenSIPS Developer

On 03.09.2013 12:14, Milos Mosovsky wrote:
> Hello , im using opensips + rtp proxy , and someone calls are very
> jitter , and lagging, for example i dont hear nothing for 5 seconds ,
> and then i hear in 1 sceond all from past 5 seonds. Sometimes sound is
> crystal clear.
> Im testing it on high speed connection , so it should no be problem in devices.
> Can be problem in codec which are used by devices? Can i somehow
> change or force codecs to better on my server?
> I know SIP is only sip proxy so it cant manipulate codec but can RTP
> proxy manipulate codecs? Thanks a lot.
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