Hey,<div>Can you send in the SIP/SDP invites. I suspect the codecs issue here.</div><div>--</div><div>Regards,</div><div>Sammy<br><div><br><br><div class="gmail_quote">On Sun, Oct 9, 2011 at 8:57 AM, Austin Einter <span dir="ltr"><<a href="mailto:austin.einter@gmail.com" target="_blank">austin.einter@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div>Hi </div>
<div>I am using Kamailio 3.1.5 . I am using RTP proxy also.</div>
<div>I have used default kamailio.cfg.sample fiile , and just added line #!define WITH_NAT.</div>
<div> </div>
<div>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.</div>
<div> </div>
<div>With this I found, call is established, all signalling and media packets are passing through kamailio / rtp-proxy.</div>
<div>So far so good.</div>
<div> </div>
<div>One way audio stream (from called party to calling party) quality is good.</div>
<div>The other audio stream (from calling party to called party is very bad.</div>
<div> </div>
<div>Did anybody face this issue? Please help me to sort out this issue audio quality issue.</div>
<div> </div>
<div>Regards</div>
<div>Austin</div><font color="#888888">
<div> </div>
<div> </div>
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