RE: [Asterisk-Users] P2P RTP without SIP re-invites
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:[EMAIL PROTECTED] Sent: 03 February 2004 07:39 To: [EMAIL PROTECTED] 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] P2P RTP without SIP re-invites
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:[EMAIL PROTECTED] Sent: 31 January 2004 13:20 To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] P2P RTP without SIP re-invites
Hi Adam, could you share your clustering setup? David -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Low, Adam Verzonden: maandag 2 februari 2004 12:11 Aan: '[EMAIL PROTECTED]' Onderwerp: RE: [Asterisk-Users] P2P RTP without SIP re-invites 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:[EMAIL PROTECTED] Sent: 31 January 2004 13:20 To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] P2P RTP without SIP re-invites
Let's go through how SIP works in Asterisk compared with a SIP Proxy. Remember that Asterisk is not designed to be a SIP Proxy, it's designed to be a Multi-VOIP and PSTN PBX, a quite complicated task. (I'm not going into all details (ACK, TRYING, RINGING etc)) We have two SIP users, Alice and Bob. Alice calls BOB, both connected to Asterisk: * Alice's UA sends an INVITE to [EMAIL PROTECTED] * Asterisk checks if bob is a valid user reachable within the context allowed by Alice's account * Asterisk answers the SIP call from Alice * Asterisk initiates another SIP call to Bob's UA with a NEW Invite * When Bob answers, Asterisk bridges the streams, performing codec conversion if necessary In this scenario, we now have two different SIP dialogues (two separate SIP calls) If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it goes directly from Alice to Bob Not all UAs support a re-INVITE and in public scenarios, a lot of UAs have broken NAT support so the RTP media stream stays with Asterisk. The benefit of this is that Asterisk acting as a user agent server (Alice) and client (bob) can send early media to Alice, connect to voicemail or another extension than Bob if Bob had issued a forward - maybe a H.323 connection or PSTN connection somewhere. With a SIP proxy we have the following scenario: * Alice's UA sends an INVITE to [EMAIL PROTECTED] * The proxy responsible for thte domain receives this and looks up bob in a user location or alias table * The proxy *FORWARDS* the same INVITE to [EMAIL PROTECTED], maybe several different locations * When Bob answers somewhere, the proxy cancels the call to the non-answering locations and forwards the OK to Alice * Alice ACKs the OK to bob and the call is UP In this scenario, there's only one SIP dialogue, between Alice and Bob with the proxy in the middle of signalling, but acting as a proxy and not as a user agent (the proxy can't and should not answer or originate calls). --- So, back to the original question, in a large installation (many users) - how do you off-load Asterisk? There's no single truth here, but here's my opinion: * If you are all on the same internal network, make sure the SIP phones support re-invites and use that. * If you have users all over the Internet, use a SIP proxy as a front-end to Asterisk You will still be forced to handle a lot of RTP streams (because of NAT), but can distribute that over a SIP-proxy network with SRV records, DNS round-robin techniques or forcing the users to register with different proxies. There's been a couple of suggestions that we should make Asterisk a good SIP proxy. If you spend some time learning to understand Asterisk's architecture, you'll also understand that this would not really work. I'm not saying the SIP channel can't be improved, I'm just saying that it has to work with the rest of Asterisk's architecture. I might be totally wrong, but my gut feeling is that Asterisk in combination with a separate SIP proxy is a very powerful solution. Clustering Asterisk servers somehow is also a good approach, but not here yet for SIP. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] P2P RTP without SIP re-invites
Hi. If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it goes directly from Alice to Bob Not all UAs support a re-INVITE and in public scenarios, a lot of UAs have broken NAT support so the RTP media stream stays with Asterisk. a small correction: doesn't matter if Alice and Bob are nat'ed: if they're both nat'ed re-INVITEs are sent and RTP is transferred to go directly from Alice to Bob. Asterisk manages only the signalling on port 5060 (I'm using that environment, so it works :) ) But if only Alice OR Bob are nat'ed, the RTP is handled by * itself. Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] P2P RTP without SIP re-invites
Brancaleoni Matteo wrote: Hi. If both Alice and Bob are connected without NAT, have the same codec support and have canreinvite=yes * Asterisk send (re-)INVITEs to both, trying to get the RTP stream transferred so it goes directly from Alice to Bob Not all UAs support a re-INVITE and in public scenarios, a lot of UAs have broken NAT support so the RTP media stream stays with Asterisk. a small correction: doesn't matter if Alice and Bob are nat'ed: if they're both nat'ed re-INVITEs are sent and RTP is transferred to go directly from Alice to Bob. Asterisk manages only the signalling on port 5060 (I'm using that environment, so it works :) ) But if only Alice OR Bob are nat'ed, the RTP is handled by * itself. 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. If someone made a solution that * Compared the inside address AND the outside (NAT public IP) * If they are similar (NAT from the same network and public IP equals), connect the RDP streams from inside NAT to inside NAT However, with STUN, the calee or the caller might not present the inside IP address and therefore this will not be possible at all... Better to have an outbound SIP proxy that could make this happen. Or? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] P2P RTP without SIP re-invites
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.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
