[OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK

[Marco Hierl]

Hi Patrik,

thanks for this idea!

I did not say clear enough: I’m afraid that anybody can cheat us. My intention is to assure that our interconnection partners (or their customers) do not have the possibility to make a conversation without being charged.
Sending the indication “a:sendonly” only means, that the client is told not to send RTP, but IF it send RTP anyway then the RTPproxy leads in on to the callee. So, it is not in my hands then!

Best regards from Hamburg
  Marco




Von: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] Im Auftrag von Patrick Wakano
Gesendet: Donnerstag, 22. Januar 2015 11:16
An: OpenSIPS users mailling list
Betreff: Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK

Have you tried to insert a "a:sendonly" line in your SDP body when sending it to the caller?
If the client receives such line it should not send media...
Then in the 200Ok you can put an "a:sendrecv" line to establish full media path!
It's just an idea, I'm not sure if it will really work...

Patrick


On Thu, Jan 22, 2015 at 6:51 AM, Marco Hierl <marco.hierl at mrnetgroup.com<mailto:marco.hierl at mrnetgroup.com>> wrote:
Hi Răzvan,

Ok, thanks for your answer!
Unfortunately we are offering „early media“ to our customers (call center, radio station, and other companies) and lots of them like to play a free-of-charge announcement in the beginning. But if we started to get cheated, maybe we need to go for this workaround.

But apart from that: Mostly the SDP is NOT repeated in the 200OK. Can I call rtpproxy_answer() when receiving the 200OK anyway?

Thanks and best regards
  Marco



Von: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> [mailto:users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org>] Im Auftrag von Razvan Crainea
Gesendet: Donnerstag, 22. Januar 2015 09:36
An: users at lists.opensips.org<mailto:users at lists.opensips.org>
Betreff: Re: [OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK

Hi, Marco!

From RTPProxy point of view, you can't differentiate between SIP replies, because for all of them you call the same function - rtpproxy_answer().
Now, if the client decides to send RTP for 183 (and indeed, I've seen this several times), there's not that much that you can do. Although it's kind of a hack, all I can think of is to not call rtpproxy_answer() for 180/183 and strip the body to prevent the client from sending RTP directly to the callee.
I hope this works for you.

Best regards,

Răzvan Crainea

OpenSIPS Solutions

www.opensips-solutions.com<http://www.opensips-solutions.com>
On 01/21/2015 04:07 PM, Marco Hierl wrote:
Dear all,

first of all I need to apologize that I was not able to find information about this issue although I’m sure that I’m not the first one complaining!

The caller is sending an INVITE via OpenSIPS and rtpproxy_offer() is executed, callee answers with REPLY 180 or REPLY 183 (with SDP) and rtpproxy_answer() is made. In this status it should be ok that the rtp stream from callee to caller is transferred via the rtpproxy (e.g. for announcements), but I can see that rtp stream from caller to callee is transferred too!!! This means that there can be a conversation without receiving the 200OK and what is the real problem: that means (at least for me) they can talk to each other without any charging !! A timer will stop the conversion after the a while, but this can take time.

How can I overcome this problem? How can prevent RTP to be send to the callee before REPLY 200 is received?

I can’t find any help in the RTPproxy protocol http://www.b2bua.org/wiki/RTPproxy/Protocol, nor in the rtpproxy module description in OpenSIPS.

Thanks for your ideas, and best regards
  Marco




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