<html><head><style type='text/css'>p { margin: 0; }</style><style type='text/css'>body { font-family: 'Times New Roman'; font-size: 12pt; color: #000000}</style></head><body>Hi Ian,<br><br>Sincere thanks for your reply.<br><br>I am attaching a patch herewith. Please let me know if you are referring to the same and this patch did not work for me. If it works for you can you pl let me know wat are changes that has to be made in the opensip,cfg file?<br><br>Else if patch is different, pl send me the patch.<br><br>Regards,<br>Arjun<br><br>----- Original Message -----<br>From: "Ian Buckner" <ian.buckner1@googlemail.com><br>To: "OpenSIPS users mailling list" <users@lists.opensips.org><br>Sent: Monday, June 11, 2012 3:48:40 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi<br>Subject: Re: [OpenSIPS-Users] No Voice Comm in Conference call<br><br>Hi Arjun,<div><br></div><div>Regarding rtpproxy - if it makes your life easier, there is a very simple patch for the rtpproxy source which allows you to specify an advertised IP address on the command line this will be returned to OpenSips and inserted into SDP rather than the listening address. I used this last week for a server behind NAT and it worked perfectly.</div><div><br></div><div>If you want to mail me I'll happily send you a patched tarball of source.</div><div><br></div><div><br></div><div>best,</div><div><br></div><div>Ian<br><div><br></div><div><br><div><br></div><div><br></div><div><br><div><div>On 11 Jun 2012, at 11:14, Bogdan-Andrei Iancu wrote:</div><br class="Apple-interchange-newline"><blockquote>
<div>
<tt>Hi Arjun,<br>
<br>
If you have no audio at all for the call to Conf Server, you need
to check the signaling, particularity speaking the SDP part to be
sure you inserted RTPProxy correctly between UAC and Conf Server.<br>
<br>
So, make a SIP capture for the call to Conf and check if the IPs
in SDP are correct.<br>
<br>
Regards,<br>
</tt>
<pre class="moz-signature">Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
<a class="moz-txt-link-freetext" href="http://www.opensips-solutions.com/" target="_blank">http://www.opensips-solutions.com</a></pre>
<br>
On 06/07/2012 05:46 PM, Arjun Shankar K S wrote:
<blockquote cite="mid:33240807.46231339080401147.JavaMail.root@mail.vlinksys.com">
<style>p { margin: 0; }</style>
<style>body { font-family: 'Times New Roman'; font-size: 12pt; color: #000000}</style>Hi
All,<br>
<br>
Greetings to everyone !!!<br>
<br>
I have set up opensips and RTP Proxy in two different hosts since
I have opensips in a Natted environment where RTP Proxy refused to
budge.<br>
<br>
Now I have installed RTP Proxy in a direct public IP. Normal calls
between 2 Client is working great !!<br>
<br>
During conference call, the calls get connected but there is no
voice communication between any of them and soon the client who
was connected last, gets disconnected.<br>
<br>
I could not find much support regarding this issue. Any support is
sincerely appreciated.<br>
<br>
In my opensips.cfg, I have made the following config changes for
RTP in different host,<br>
<br>
<span style="font-family: Courier,Courier New,mono; font-style: italic;">------Nat Params-------------</span><br style="font-family: Courier,Courier New,mono; font-style: italic;">
<span style="font-family: Courier,Courier New,mono; font-style: italic;">modparam("usrloc","nat_bflag", 6)</span><br style="font-family: Courier,Courier New,mono; font-style: italic;">
<span style="font-family: Courier,Courier New,mono; font-style: italic;">modparam("nathelper","rtpproxy_sock",
"udp:rtp_proxy_publicIP:7890")</span><br style="font-family: Courier,Courier New,mono; font-style: italic;">
<span style="font-family: Courier,Courier New,mono; font-style: italic;">modparam("nathelper","natping_interval", 30)</span><br style="font-family: Courier,Courier New,mono; font-style: italic;">
<span style="font-family: Courier,Courier New,mono; font-style: italic;">modparam("nathelper","ping_nated_only", 0)</span><br style="font-family: Courier,Courier New,mono; font-style: italic;">
<span style="font-family: Courier,Courier New,mono; font-style: italic;">modparam("nathelper","sipping_bflag", 7)</span><br style="font-family: Courier,Courier New,mono; font-style: italic;">
<span style="font-family: Courier,Courier New,mono; font-style: italic;">modparam("nathelper","sipping_from",
<a class="moz-txt-link-rfc2396E" href="about:blank" target="_blank">"sip:pinger@PROXY_IP"</a>)</span><br style="font-family: Courier,Courier New,mono; font-style: italic;">
<span style="font-family: Courier,Courier New,mono; font-style: italic;">modparam("registrar","received_avp", "$avp(i:42)")</span><br style="font-family: Courier,Courier New,mono; font-style: italic;">
<span style="font-family: Courier,Courier New,mono; font-style: italic;">modparam("nathelper","received_avp", "$avp(i:42)")<br>
<br>
</span><span style="font-family: Times New Roman,Times,serif;">I
am </span><span style="font-family: Courier,Courier New,mono;"><span style="font-family: Times New Roman,Times,serif;">running my
RTP Proxy using the following command,</span></span><span style="font-family: Courier,Courier New,mono; font-style: italic;"><br>
<br>
./rtpproxy -l </span><span style="font-family: Courier,Courier New,mono; font-style: italic;">rtp_proxy_publicIP</span><span style="font-family: Courier,Courier New,mono; font-style: italic;"> -s udp:*:7890 -F<br>
<br>
</span><br>
Thanks,<br>
Arjun<br>
<pre><fieldset class="mimeAttachmentHeader"></fieldset>
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