No. You can't route the RTP and RTCP traffic to Kamailio, by definition.

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.

On 26 March 2014 04:21:08 GMT-04:00, Cock Ootec <> wrote:
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.

On Wed, Mar 26, 2014 at 8:56 AM, Olle E. Johansson <> wrote:

On 26 Mar 2014, at 08:44, Cock Ootec <> wrote:

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?
Yes, that is how the SIP protocol works. Please update yourself on the protocol which Kamailio is built around.

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?

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.
You can modify as much as you want.


On Wed, Mar 26, 2014 at 7:41 AM, Olle E. Johansson <> wrote:

On 26 Mar 2014, at 01:06, Cock Ootec <> wrote:

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 topoh?

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.

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.

Before you modify software, you need to understand the architecture :-)


In topoh there are two eventsĀ SREV_NET_DATA_IN and SREV_NET_DATA_OUT but I didn't be able to capture any RTP packets by them.

Thanks in advance for any help or useful information.
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