Hey, Can you send in the SIP/SDP invites. I suspect the codecs issue here. -- Regards, Sammy
On Sun, Oct 9, 2011 at 8:57 AM, Austin Einter <austin.ein...@gmail.com>wrote: > Hi > I am using Kamailio 3.1.5 . I am using RTP proxy also. > I have used default kamailio.cfg.sample fiile , and just added line > #!define WITH_NAT. > > I have another Main proxy. I wanted all my signalling and media packets > should just pass through machine where Kamailio and RTP proxy are running. > > With this I found, call is established, all signalling and media packets > are passing through kamailio / rtp-proxy. > So far so good. > > One way audio stream (from called party to calling party) quality is good. > The other audio stream (from calling party to called party is very bad. > > Did anybody face this issue? Please help me to sort out this issue audio > quality issue. > > Regards > Austin > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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