<html><head></head><body>No. You can't route the RTP and RTCP traffic to Kamailio, by definition. <br>
<br>
You keep asking questions that betray a lack of basic understanding of SIP network elements. I think you should take Olle's suggestion and learn how it works. <br>
<br><br><div class="gmail_quote">On 26 March 2014 04:21:08 GMT-04:00, Cock Ootec <cockootec@gmail.com> wrote:<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div dir="ltr">Ok so if I explicitly route all my VoIP traffic (SIP, RTP, RTCP) to Kamailio I can distinguish the streams (parse packets, edit packets) and for example forward these streams to different ports? Thats perfect. Thanks for all your responses they helped much.</div>
<div class="gmail_extra"><br /><br /><div class="gmail_quote">On Wed, Mar 26, 2014 at 8:56 AM, Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>></span> wrote:<br /><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="word-wrap:break-word"><br /><div><div class=""><div>On 26 Mar 2014, at 08:44, Cock Ootec <<a href="mailto:cockootec@gmail.com" target="_blank">cockootec@gmail.com</a>> wrote:</div><br /><blockquote type="cite">
<div dir="ltr">Thanks for your quick reply. So, Kamailio does nothing with RTP/RTCP packets? I think I understand (now when I think about it) - Kamailio only handles SIP messages by which in nested SDP two endpoints negotiate the stream where media will go through? </div>
</blockquote></div>Yes, that is how the SIP protocol works. Please update yourself on the protocol which Kamailio is built around.</div><div><div class=""><br /><blockquote type="cite"><div dir="ltr">So RTP/RTCP media stream flow directly between two UA endpoints and Kamailio has nothing to do with handling of these packets. Could you, please confirm my thoughts?</div>
</blockquote></div>Yes.<div class=""><br /><blockquote type="cite"><div dir="ltr"><div>
<br /></div><div>All right but what if for example we have special UA that sends to Kamailio specially modified packet (non standard SIP). In my extension of topoh module I have sanity_checks disabled so will Kamailio check this packet before my module and drops it or I can receive, modify and forward this packet? I mean modify this packet to standard SIP packet and forward it to another UA. I am just asking theoretically because in the moment I cant try this.</div>
</div></blockquote></div>You can modify as much as you want.</div><span class="HOEnZb"><font color="#888888"></font><div><font color="#888888"><br /></font></div></span><div><span class="HOEnZb"><font color="#888888">/O</font></span><div class=""><br /><blockquote type="cite">
<div dir="ltr">
<div><br /></div></div><div class="gmail_extra"><br /><br /><div class="gmail_quote">On Wed, Mar 26, 2014 at 7:41 AM, Olle E. Johansson <span dir="ltr"><<a href="mailto:oej@edvina.net" target="_blank">oej@edvina.net</a>></span> wrote:<br />
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word"><br /><div><div><div>On 26 Mar 2014, at 01:06, Cock Ootec <<a href="mailto:cockootec@gmail.com" target="_blank">cockootec@gmail.com</a>> wrote:</div>
<br /><blockquote type="cite"><div dir="ltr">Hi,<div>I'd like to develop module for Kamailio which will be working with RTP/RTCP packets. Is there any way to capture and edit RTP packet in module of Kamailio for example in module extended from <b>topoh</b>?</div>
<div><br /></div></div></blockquote></div><div>Kamailio in itself is a SIP server, often acting as a SIP proxy. The SIP protocol doesn't handle media, it facilitates setup and management of a media session. Adding a module for handling media in Kamailio doesn't really make any sense.</div>
<div><br /></div><div>We do have modules that talk to external media servers. Look into those - like rtpproxy. The Kamailio module itself does not handle media, but communicates with the other server that in fact manages media relaying.</div>
<div><br /></div><div>Before you modify software, you need to understand the architecture :-)</div><div><br /></div><div>/O</div><div><br /></div><br /><blockquote type="cite"><div><div dir="ltr"><div>In topoh there are two events <span style="color:rgb(51,51,51);font-family:Consolas,'Liberation Mono',Courier,monospace;font-size:12px;line-height:18px;white-space:pre-wrap">SREV_NET_DATA_IN and </span><span style="color:rgb(51,51,51);font-family:Consolas,'Liberation Mono',Courier,monospace;font-size:12px;line-height:18px;white-space:pre-wrap">SREV_NET_DATA_OUT </span>but I didn't be able to capture any RTP packets by them.</div>
<div><br /></div><div>Thanks in advance for any help or useful information.</div></div></div>
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