From jbrower at signalogic.com Tue Oct 20 16:25:42 2009 From: jbrower at signalogic.com (Jeff Brower) Date: Tue, 20 Oct 2009 18:25:42 -0500 (CDT) Subject: [RTPproxy Users] (no subject) Message-ID: <2893.64.219.188.225.1256081142.squirrel@office.signalogic.com> All- On a reasonably fast server, say quad-core, what is the approx maximum number of G711 IP calls when using OpenSIPS (OpenSER) and rtpproxy? What if rtpproxy runs on a second server? No echo can, no packet concealment, etc... just G711. -Jeff From mayamatakeshi at gmail.com Wed Oct 21 08:13:38 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Thu, 22 Oct 2009 00:13:38 +0900 Subject: [RTPproxy Users] RTPProxy sending RTP packets before receiving initial packet. Message-ID: <15b9404e0910210813h588f3f49iea1d3a1d33859672@mail.gmail.com> Hello, we are running RTPPRoxy 1.1 with Kamailio. After a call is set up, I can see rtpproxy sending 24 RTP packets to the private ip and port advertised in the SDP of the caller (like 192.168.1.2:60000). This doesn't cause any problem to the call, as less then one second after this, rtpproxy starts to send the packets to the global IP addresses. However, why would be rtpproxy doing this? It doesn't seem to be according to the HOWITWORKS explanation in the man page. Does rtpproxy receive this SDP info from SER? The problem is that we have those addresses (192.168.1.*) in our network, so this causes several "icmp destination unreachable" packets to be sent back and this is flooding our network. And I don't see these private ports sending any packets to rtpproxy ports to make it behave as such. Regards, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.rtpproxy.org/pipermail/users/attachments/20091022/438ec7e4/attachment.html From jbrower at signalogic.com Thu Oct 22 15:57:03 2009 From: jbrower at signalogic.com (Jeff Brower) Date: Thu, 22 Oct 2009 17:57:03 -0500 (CDT) Subject: [RTPproxy Users] basic SIP forwarding with Asterisk Message-ID: <3723.64.219.188.225.1256252223.squirrel@office.signalogic.com> All- Can we use Asterisk combined with Kamailio as follows: __________ ___________ | | | | SIP ___| |___ SIP ___| Kamailio |___ SIP | | | rtpproxy | | Asterisk | | | | | | | | | RTP ___| |___ RTP ___| DSP card |___ RTP (G711) |__________| (G711) |___________| (G729, G723, GSM-AMR, EVRC) We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is whether we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and tear-down and/or when media attributes for the call change? We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use Asterisk, but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't support (or at least doesn't support at higher capacity or without going unstable). Thanks. -Jeff From klaus.mailinglists at pernau.at Fri Oct 23 06:15:56 2009 From: klaus.mailinglists at pernau.at (Klaus Darilion) Date: Fri, 23 Oct 2009 15:15:56 +0200 Subject: [RTPproxy Users] [Kamailio-Users] basic SIP forwarding with Asterisk In-Reply-To: <3723.64.219.188.225.1256252223.squirrel@office.signalogic.com> References: <3723.64.219.188.225.1256252223.squirrel@office.signalogic.com> Message-ID: <4AE1AC8C.7000505@pernau.at> Hi Jeff! So you want to do transcoding in rtpproxy using a DSP card? I do not know - better ask on the rtpproxy mailing list (or Maxim directly - I think he has a non-open source solution). Anyway - why not do the transcoding in Asterisk? regards klaus Jeff Brower schrieb: > All- > > Can we use Asterisk combined with Kamailio as follows: > > __________ ___________ > | | | | > SIP ___| |___ SIP ___| Kamailio |___ SIP > | | | rtpproxy | > | Asterisk | | | | > | | | | | > RTP ___| |___ RTP ___| DSP card |___ RTP > (G711) |__________| (G711) |___________| (G729, > G723, > GSM-AMR, > EVRC) > > We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is whether > we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and > tear-down and/or when media attributes for the call change? > > We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use Asterisk, > but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't > support (or at least doesn't support at higher capacity or without going unstable). > > Thanks. > > -Jeff > > > _______________________________________________ > Kamailio (OpenSER) - Users mailing list > Users at lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > http://lists.openser-project.org/cgi-bin/mailman/listinfo/users From jbrower at signalogic.com Fri Oct 23 08:01:41 2009 From: jbrower at signalogic.com (Jeff Brower) Date: Fri, 23 Oct 2009 10:01:41 -0500 (CDT) Subject: [RTPproxy Users] [Kamailio-Users] basic SIP forwarding with Asterisk In-Reply-To: <4AE1AC8C.7000505@pernau.at> References: <3723.64.219.188.225.1256252223.squirrel@office.signalogic.com> <4AE1AC8C.7000505@pernau.at> Message-ID: <1915.71.164.240.84.1256310101.squirrel@office.signalogic.com> Klaus- > So you want to do transcoding in rtpproxy using a DSP card? I do not > know - better ask on the rtpproxy mailing list (or Maxim directly - I > think he has a non-open source solution). Ya we have -- and it works, no problem. We've tested already with Kamailio + rtpproxy. > Anyway - why not do the transcoding in Asterisk? Because Asterisk is too limited. It can't do enough channels for G729, and doesn't have good options for codecs like EVRC and GSM-AMR. But anyway my question is about SIP with Kamailio + Asterisk, not RTP. Is there a way that Kamailio can "pass thru" SIP messages from Asterisk? Or does each call have to be relayed; i.e Asterisk sets up a call to Kamailio, then Kamalio sets up a call to the endpoint? I know that we can get it to work one way or another, but I'm worried about channel capacity if both Asterisk and Kamailio run on the same server. "Duplicating" calls does not seem efficient. -Jeff > Jeff Brower schrieb: >> All- >> >> Can we use Asterisk combined with Kamailio as follows: >> >> __________ ___________ >> | | | | >> SIP ___| |___ SIP ___| Kamailio |___ SIP >> | | | rtpproxy | >> | Asterisk | | | | >> | | | | | >> RTP ___| |___ RTP ___| DSP card |___ RTP >> (G711) |__________| (G711) |___________| (G729, >> G723, >> GSM-AMR, >> EVRC) >> >> We've already implemented an rtpproxy interface to the DSP card, which has its own GbE port. Our question is >> whether >> we can we perform some type of basic SIP forwarding or "SIP pass-thru", but still invoke rtpproxy for call setup and >> tear-down and/or when media attributes for the call change? >> >> We're getting a lot of requests from customers who -- for whatever reasons -- need to use or continue to use >> Asterisk, >> but need to also add transcoding, ec, encryption, and other compute-intensive requirements that Asterisk doesn't >> support (or at least doesn't support at higher capacity or without going unstable). >> >> Thanks. >> >> -Jeff >> >> >> _______________________________________________ >> Kamailio (OpenSER) - Users mailing list >> Users at lists.kamailio.org >> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >> http://lists.openser-project.org/cgi-bin/mailman/listinfo/users > From klaus.mailinglists at pernau.at Fri Oct 23 08:39:00 2009 From: klaus.mailinglists at pernau.at (Klaus Darilion) Date: Fri, 23 Oct 2009 17:39:00 +0200 Subject: [RTPproxy Users] [Kamailio-Users] basic SIP forwarding with Asterisk In-Reply-To: <1915.71.164.240.84.1256310101.squirrel@office.signalogic.com> References: <3723.64.219.188.225.1256252223.squirrel@office.signalogic.com> <4AE1AC8C.7000505@pernau.at> <1915.71.164.240.84.1256310101.squirrel@office.signalogic.com> Message-ID: <4AE1CE14.8010507@pernau.at> Jeff Brower schrieb: > Klaus- > >> So you want to do transcoding in rtpproxy using a DSP card? I do not >> know - better ask on the rtpproxy mailing list (or Maxim directly - I >> think he has a non-open source solution). > > Ya we have -- and it works, no problem. We've tested already with Kamailio + rtpproxy. > >> Anyway - why not do the transcoding in Asterisk? > > Because Asterisk is too limited. It can't do enough channels for G729, and doesn't have good options for codecs like > EVRC and GSM-AMR. > > But anyway my question is about SIP with Kamailio + Asterisk, not RTP. Is there a way that Kamailio can "pass thru" > SIP messages from Asterisk? Or does each call have to be relayed; i.e Asterisk sets up a call to Kamailio, then > Kamalio sets up a call to the endpoint? What is the difference between "pass thru" and relaying? Kamailio is a proxy, that means it receives a SIP message from somebody and sends it (slighty modified) to somebody. This forwarding can be done stateless or transaction statefull. Unless you use the dialog module, Kamailio does not care about "set-up" calls, so if you have 100000000 millions if calls set up, Kamailio is idle as it only cares about transactions (INVITE, BYE ...), not about ongoing calls. regards klaus From jbrower at signalogic.com Fri Oct 23 11:10:42 2009 From: jbrower at signalogic.com (Jeff Brower) Date: Fri, 23 Oct 2009 13:10:42 -0500 (CDT) Subject: [RTPproxy Users] [Kamailio-Users] basic SIP forwarding with Asterisk In-Reply-To: <4AE1CE14.8010507@pernau.at> References: <3723.64.219.188.225.1256252223.squirrel@office.signalogic.com> <4AE1AC8C.7000505@pernau.at> <1915.71.164.240.84.1256310101.squirrel@office.signalogic.com> <4AE1CE14.8010507@pernau.at> Message-ID: <4951.64.219.188.225.1256321442.squirrel@office.signalogic.com> Klaus- > Jeff Brower schrieb: >> Klaus- >> >>> So you want to do transcoding in rtpproxy using a DSP card? I do not >>> know - better ask on the rtpproxy mailing list (or Maxim directly - I >>> think he has a non-open source solution). >> >> Ya we have -- and it works, no problem. We've tested already with Kamailio + rtpproxy. >> >>> Anyway - why not do the transcoding in Asterisk? >> >> Because Asterisk is too limited. It can't do enough channels for G729, and doesn't have good options for codecs >> like >> EVRC and GSM-AMR. >> >> But anyway my question is about SIP with Kamailio + Asterisk, not RTP. Is there a way that Kamailio can "pass thru" >> SIP messages from Asterisk? Or does each call have to be relayed; i.e Asterisk sets up a call to Kamailio, then >> Kamalio sets up a call to the endpoint? > > What is the difference between "pass thru" and relaying? Kamailio is a > proxy, that means it receives a SIP message from somebody and sends it > (slighty modified) to somebody. This forwarding can be done stateless or > transaction statefull. > > Unless you use the dialog module, Kamailio does not care about "set-up" > calls, so if you have 100000000 millions if calls set up, Kamailio is > idle as it only cares about transactions (INVITE, BYE ...), not about > ongoing calls. Thanks for your reply. Yes you're right... I think we just need to try it stateless and measure the performance. -Jeff From mayamatakeshi at gmail.com Sat Oct 31 18:42:04 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sun, 1 Nov 2009 10:42:04 +0900 Subject: [RTPproxy Users] RTPProxy sending RTP packets before receiving initial packet. In-Reply-To: <15b9404e0910210813h588f3f49iea1d3a1d33859672@mail.gmail.com> References: <15b9404e0910210813h588f3f49iea1d3a1d33859672@mail.gmail.com> Message-ID: <15b9404e0910311842g5ecac831g85e44d0b041fabb@mail.gmail.com> On Thu, Oct 22, 2009 at 12:13 AM, mayamatakeshi wrote: > Hello, > we are running RTPPRoxy 1.1 with Kamailio. > After a call is set up, I can see rtpproxy sending 24 RTP packets to the > private ip and port advertised in the SDP of the caller (like > 192.168.1.2:60000). > This doesn't cause any problem to the call, as less then one second after > this, rtpproxy starts to send the packets to the global IP addresses. > However, why would be rtpproxy doing this? It doesn't seem to be according > to the HOWITWORKS explanation in the man page. > Does rtpproxy receive this SDP info from SER? > The problem is that we have those addresses (192.168.1.*) in our network, > so this causes several "icmp destination unreachable" packets to be sent > back and this is flooding our network. > And I don't see these private ports sending any packets to rtpproxy ports > to make it behave as such. > Trying to solve problems with my kamailio installation I think I got the reason rtprproxy does this: this is necessary to avoid deadlock in case there are other instances of rtpproxy handling the call. rtpproxy cannot just sit and wait for the initial packets to come from both ends: if it does, RTP will not start on a chain of rtpproxies because no one will start the transmission. br, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.rtpproxy.org/pipermail/users/attachments/20091101/c498fe64/attachment.html