I'm confronted with an issue that I am sure many others are too with Asterisk and scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a large volume of simultaneous calls but have the feeling that the hardware requirements to handle large volumes of RTP streams would be too vast. So with that assumption I imagine a platform that would not get involved with the actual encoding/decoding of the RTP stream ensuring that only the SIP client's on each end of the call deal with RTP encoding with their dedicated DSP hardware. There is an alternative in mind that maybe I could utilise some old Dialogic DSP cards that we have but I suspect trying to get these working with Asterisk would be a lot of programming work that I probably couldn't manage, maybe I'm wrong ? The SIP RE-INVITE mechanism is useful but I find problems when SIP clients are NAT'd (specifically SIP breaks and calls are not torn down correctly) and of course you lose a lot of monitoring (CDR's, etc.)and management capabilities provided by Asterisk when it is in the SIP signalling path. I vaguely remember previous discussions on this and even a patch but I am unable to find anything in the archives, does anybody have any info on that ? The conclusion I have come to is that I would try and patch the Asterisk code. The idea being that when the RTP parameters are negotiated that Asterisk would pass through the source address/port from each SIP client causing them to talk RTP directly. I intend to begin work on this this weekend but am I hoping that maybe somebody else has already achieved what I desire, anybody ? Rgds, Adam ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
Low, Adam wrote:>I'm confronted with an issue that I am sure many others are too with Asterisk and scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a large volume of simultaneous calls but have the feeling that the hardware requirements to handle large volumes of RTP streams would be too vast. > >So with that assumption I imagine a platform that would not get involved with the actual encoding/decoding of the RTP stream ensuring that only the SIP client's on each end of the call deal with RTP encoding with their dedicated DSP hardware. There is an alternative in mind that maybe I could utilise some old Dialogic DSP cards that we have but I suspect trying to get these working with Asterisk would be a lot of programming work that I probably couldn't manage, maybe I'm wrong ? > >The SIP RE-INVITE mechanism is useful but I find problems when SIP clients are NAT'd (specifically SIP breaks and calls are not torn down correctly) and of course you lose a lot of monitoring (CDR's, etc.)and management capabilities provided by Asterisk when it is in the SIP signalling path. > >I vaguely remember previous discussions on this and even a patch but I am unable to find anything in the archives, does anybody have any info on that ? > >The conclusion I have come to is that I would try and patch the Asterisk code. The idea being that when the RTP parameters are negotiated that Asterisk would pass through the source address/port from each SIP client causing them to talk RTP directly. I intend to begin work on this this weekend but am I hoping that maybe somebody else has already achieved what I desire, anybody ? > >Rgds, >Adam > > > > > >Asterisk single system scaling is an issue that I have been thinking about as well, and wondering about ways to cluster multiple Asterisk servers together to act as a unified system.. So far I haven't really got anywhere becasue everytjing I have thought of has been a problem most related to RTP.. Of course remember that the RTP is not really that much of a problem (apart from the bandwidth usage) when both the UA's are using the same codec.. Asterisk will simply switch the encoded voice traffic.. I am sure some clever person will come up with an answer but whether or not they share it is another question.. later..
Apologies for the belated reply but I've spent the weekend fighting DDoS attacks against Superbowl sites ... )c; Ok, well I am not sure what went wrong with previous testing but I have tried this again with Cisco 7940's and Cisco AS5300's and indeed the RTP stream flows directly between end-points retaining SIP signalling via Asterisk. This is exactly the operation I had hoped for. I had previously tested with my home 7940 which it behind NAT without success and so will re-test this this evening. Thanks for all the responses and related discussion on clustering Asterisk, thanks to those I now have a running cluster of 3 Asterisk servers each with mirrored sip.conf and extensions.conf built dynamically from a MySQL backend database. Rgds, Adam -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: 31 January 2004 13:20 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] P2P RTP without SIP re-invites hi> > I guess this would work if both Alice and Bob were NAT'ed on the inside of the same > NAT box. The problem is that if Alice and Bob both have NAT=yes and CANREINVITE=yes > and they're on separate NAT'ed networks, the call is broken. So it's a dangerous > configuration.nope. I have a public * server (beta server for a free VoIP service), on a public IP. and some sip phones around , like one in my home, behind nat, one in my office (another nat) and some others at my coworkers home... all behind nat. and are different nat box, do you agree? that works ok, I have RTP passing directly from one endpoint to the other... no RTP on the public * server. No stun is used. The phones are budgetones in this case. All are configured with nat=yes on asterisk side. or I missing something? -- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmegi Srl _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person
Several people have requested more information on my cluster setup, I'll try to put something together today but things are very busy here at the moment ... but keep an eye for a mail today ... -----Original Message----- From: David Luyens [mailto:d@vt4.net] Sent: 03 February 2004 07:39 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] P2P RTP without SIP re-invites Hi Adam, could you share your clustering setup? David ********* DISCLAIMER ********* This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person