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

[Patrick Wakano]

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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