<div dir="ltr">Dear Khoa,<div><br></div><div>As Daniel stated you need to see if your SIP phones are able to sense the change in its network parameters and trigger a Re-INVITE to Kamailio with new SDP to handle the audio. </div>
<div><br></div><div>That's very important to do because once RTPproxy allocates the ports it can't just start sending RTPs to the new A2 network IP on its own OR start receiving RTPs from new IP to its already allocated port (that'll mean anyone can send RTPs to an allocated port and insert media into someone else's call).</div>
<div><br></div><div>Please do share the network topology where the first network transition worked, possibly it's Public IP remained the same and maybe your internal network handled that somehow(NAT/PAT) !??</div><div>
<br></div><div>Now once the Re-INVITES are exchanged only then RTPproxy will be explored to see if it handles the transparent Handover/updates or not. </div><div><br></div><div>BR,</div><div>Sammy</div><div><br></div><div>
<br></div><div><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Aug 14, 2013 at 7:03 AM, Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@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 text="#000000" bgcolor="#FFFFFF">
Kamailio does not send any command to RTPProxy unless it handles
some SIP messages, the U and L commands are typically for INVITEs
and 200ok.<br>
<br>
Have you looked at sip traffic? You can run ngrep on kamailio
server:<br>
<br>
ngrep -d any -qt -W byline port 5060<br>
<br>
Cheers,<br>
Daniel<div><div class="h5"><br>
<br>
<div>On 8/14/13 1:40 PM, Khoa Pham wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">I think it is related to so called IP address
filling and trusted IP</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Wed, Aug 14, 2013 at 4:09 PM, Khoa
Pham <span dir="ltr"><<a href="mailto:onmyway133@gmail.com" target="_blank">onmyway133@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 dir="ltr">Hi Daniel,
<div><br>
</div>
<div>My clients don't do anything when IP change
occurs.From what I inspect, it is because of rtpproxy
does not accept the 2nd IP change.</div>
<div>The the rtpproxy protocol document <a href="http://www.rtpproxy.org/wiki/RTPproxy/Protocol" target="_blank">http://www.rtpproxy.org/wiki/RTPproxy/Protocol</a>,
the Update and Lookup command have [arg] parameters. </div>
<div><span style="font-size:13px;font-family:monospace">U[args]
callid addr port from_tag [to_tag [notify_socket
[notify_args]]]</span><br>
</div>
<div><span style="font-size:13px;font-family:monospace">L[args]
callid addr port from_tag to_tag</span><span style="font-size:13px;font-family:monospace"><br>
</span></div>
<div><span style="font-size:13px;font-family:monospace"><br>
</span></div>
<div><font face="monospace" color="#000000">I see Kamailio
often send Uc and Lc to rtpproxy. I still can't find
out what these arg mean, but maybe it's the point</font></div>
</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">
<div>
<div>On Wed, Aug 14, 2013 at 3:31 PM,
Daniel-Constantin Mierla <span dir="ltr"><<a href="mailto:miconda@gmail.com" target="_blank">miconda@gmail.com</a>></span>
wrote:<br>
</div>
</div>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div>
<div>
<div text="#000000" bgcolor="#FFFFFF"> Hello,
<div>
<div><br>
<br>
<div>On 8/13/13 5:56 AM, Khoa Pham wrote:<br>
</div>
<blockquote type="cite">
<div dir="ltr">
<p> I have SIP proxy (Kamailio) works in
conjunction with <a href="http://www.rtpproxy.org/" rel="nofollow" style="margin:0px;padding:0px;border:0px;vertical-align:baseline;background-color:transparent;color:rgb(74,107,130);text-decoration:none" target="_blank">rtpproxy</a> to
support client communication. When SIP
proxy sends command to rtpproxy to
create new session, rtpproxy will
create 2 ports (let's called them
port1 and port2). rtpproxy has 1
listen interface</p>
<p> Supposed A and B are 2 clients that
use rtpproxy to relay RTP stream, and
works fine.</p>
<blockquote>
<p style="margin:0px 0px 1em;padding:0px;border:0px;vertical-align:baseline;background-color:transparent;clear:both;word-wrap:break-word">A
<---> port1 [<strong style="margin:0px;padding:0px;border:0px;vertical-align:baseline;background-color:transparent">rtpproxy</strong>]
port2 <---> B</p>
</blockquote>
<p> Now that A loses his current
network, and enter network2 (imagine a
network handover) to become A2. In
this case, I see rtpproxy still works
fine by relaying stream between A2 and
B</p>
<blockquote>
<p style="margin:0px 0px 1em;padding:0px;border:0px;vertical-align:baseline;background-color:transparent;clear:both;word-wrap:break-word">A2
<---> port1 [<strong style="margin:0px;padding:0px;border:0px;vertical-align:baseline;background-color:transparent">rtpproxy</strong>]
port2 <---> B</p>
</blockquote>
<p> But when A2 lose his network2 and
enters network3 to become A3, rtpproxy
stills relay stream between A2 and B.
It seems that A can change his network
only once.</p>
<blockquote>
<p style="margin:0px 0px 1em;padding:0px;border:0px;vertical-align:baseline;background-color:transparent;clear:both;word-wrap:break-word">A2
<---> port1 [<strong style="margin:0px;padding:0px;border:0px;vertical-align:baseline;background-color:transparent">rtpproxy</strong>]
port2 <---> B</p>
<p style="margin:0px 0px 1em;padding:0px;border:0px;vertical-align:baseline;background-color:transparent;clear:both;word-wrap:break-word">A3</p>
</blockquote>
<p> Why did the first handover succeed?
How can I change rtpproxy behavior to
support many handovers ?</p>
</div>
</blockquote>
</div>
</div>
what I expect that happened between A and A2 is
that the client application sent a re-INVITE
with its new IP address. But then it didn't
happen when going to A3. Rtpproxy itself can do
nothing here. You should look at sip traffic to
see what happens.<br>
<br>
Cheers,<br>
Daniel<span><font color="#888888"><br>
<pre cols="72">--
Daniel-Constantin Mierla - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
<a href="http://twitter.com/#%21/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
</pre>
</font></span></div>
<br>
</div>
</div>
_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) -
sr-users mailing list<br>
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<br>
</blockquote>
</div>
<div><br>
<br clear="all">
<div><br>
</div>
-- <br>
<div dir="ltr"><span>Khoa Pham</span>
<div>HCMC University of Science<br>
<span><a href="http://www.fantageek.com" target="_blank">www.fantageek.com</a></span><br>
</div>
</div>
</div>
</div>
</blockquote>
</div>
<br>
<br clear="all">
<div><br>
</div>
-- <br>
<div dir="ltr"><span style>Khoa Pham</span>
<div>HCMC University of Science<br style>
<span style><a href="http://www.fantageek.com" target="_blank">www.fantageek.com</a></span><br>
</div>
</div>
</div>
</blockquote>
<br>
<pre cols="72">--
Daniel-Constantin Mierla - <a href="http://www.asipto.com" target="_blank">http://www.asipto.com</a>
<a href="http://twitter.com/#!/miconda" target="_blank">http://twitter.com/#!/miconda</a> - <a href="http://www.linkedin.com/in/miconda" target="_blank">http://www.linkedin.com/in/miconda</a>
</pre>
</div></div></div>
<br>_______________________________________________<br>
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list<br>
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<br></blockquote></div><br></div>