[OpenSIPS-Users] Sound jitter

[Milos Mosovsky]

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.