[OpenSIPS-Users] rtpproxy sends rtp from caller to callee before 200OK
Răzvan Crainea
razvan at opensips.org
Thu Jan 22 14:12:14 CET 2015
Hi Marco!
As Patrick suggested, adding the a:sendonly line in RTP should instruct
the caller not to send any RTP. However, if I remember correctly, I've
seen legitimate clients that still send RTP.
On a different note, they are sending RTP to a media gateway, right? And
most likely the B part will ignore all the RTP.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 01/22/2015 02:50 PM, Patrick Wakano wrote:
> 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 <mailto: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>
> [mailto: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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