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. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
