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
<[email protected] <mailto:[email protected]>> 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:*[email protected]
<mailto:[email protected]>
[mailto:[email protected]
<mailto:[email protected]>] *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
<[email protected] <mailto:[email protected]>>
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:*[email protected]
<mailto:[email protected]>
[mailto:[email protected]
<mailto:[email protected]>] *Im Auftrag von *Razvan
Crainea
*Gesendet:* Donnerstag, 22. Januar 2015 09:36
*An:* [email protected] <mailto:[email protected]>
*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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