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

Patrick Wakano pwakano at gmail.com
Thu Jan 22 13:50:41 CET 2015


Ok Marco,
Your concern is with hackers and not misuse! Really valid nowadays!

Patrick

On Thu, Jan 22, 2015 at 8:32 AM, Marco Hierl <marco.hierl at mrnetgroup.com>
wrote:

> 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>
> 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] *Im Auftrag von *Razvan Crainea
> *Gesendet:* Donnerstag, 22. Januar 2015 09:36
> *An:* 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
>
> 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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