Re: [asterisk-users] Re: NAT: RTP Path Optimization
PC When I set for Extern1/2 canreinvite=yes it works, but
PC Intern-2-Extern doesn't work because Asteisk gives out the
PC private IP-Adresses of Int1/2
Asterisk can't give out a public IP-address for Int1/2. Where
would it get one from?
Correct that it doesn't. But some kind sould could indeed code a
variety of techniques to get it, such as:
Again: My Problem is not Intern-to-Extern (NAT,Stun). My Problem is
Extern-to-Extern, that the external phones are not talking RTP
*directly* to each other. This is bad, when Asterisk is in Europe and
the Phones are in Asia.
___
Just an idea, it's completely unverified and if I missed the point
somewhat, please excuse me ;) - but maybe this approach leads to the OPs
desired result if thought through further.
I wonder wether some clever dialplan constructs couldn't help.
I'm thinking along the lines of:
[globals]
ALLOWDIRECT=
REMAININPATH=tTwW
[internalphones] - registration context of internal phones
exten = extern1,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = extern2,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = intern1,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
exten = intern2,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
[externalphones] - registration context of internal phones
exten = intern1,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = intern2,1,Dial(SIP/${EXTEN},30,${REMAINPATH})
exten = extern1,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
exten = extern2,1,Dial(SIP/${EXTEN},30,${ALLOWDIRECT})
This assumes all sip phones are set to reinvite=yes.
I expect (one of) the options to dial (tTw or W) to force asterisk to
remain in the media path. This way *only* if it's int-int or ext-ext
will it send sip reinvite, right?
I have another idea just now, but that's even weirder ... ;)
Conrad
___
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Re: [asterisk-users] Re: NAT: RTP Path Optimization
On Wednesday 31 January 2007 8:28 am, Conrad Wood wrote: This assumes all sip phones are set to reinvite=yes. I expect (one of) the options to dial (tTw or W) to force asterisk to remain in the media path. This way *only* if it's int-int or ext-ext will it send sip reinvite, right? Yes, but now you have to be careful of unintended consequences when people are trying to use IVRs. What's wrong with having two peers, one with canreinvite=no, and Dial() using the appropriate one? (I haven't been following the thread, so this may have already been discussed, and discounted.) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
On Wed, 2007-01-31 at 08:42 -0500, Andrew Kohlsmith wrote: On Wednesday 31 January 2007 8:28 am, Conrad Wood wrote: This assumes all sip phones are set to reinvite=yes. I expect (one of) the options to dial (tTw or W) to force asterisk to remain in the media path. This way *only* if it's int-int or ext-ext will it send sip reinvite, right? Yes, but now you have to be careful of unintended consequences when people are trying to use IVRs. I wouldn't take it live as is without further testing, but I guess the idea was worth adding to the thread. What's wrong with having two peers, one with canreinvite=no, and Dial() using the appropriate one? (I haven't been following the thread, so this may have already been discussed, and discounted.) That was the other idea I had - but think different Dial parameters are less problematic. For 2 peers per phone, the phones either need to have a static public IP or need to be able to register with 2 credentials (e.g. snom/cisco). Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
Thanks for this discussion! I've gotten a few ideas for better NAT handling in chan_sip3. The current way is implemented in so many installations, so it would be hard to turn it around, but in pineapple I can freely break backwards compatibility. Let me think about it for a few days, then I'll try to summarize. What's Pineapple and chan_sip3? Check http://www.codename-pineapple.org Major update coming soon, hopefully. /O ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote: PC When I set for Extern1/2 canreinvite=yes it works, but PC Intern-2-Extern doesn't work because Asteisk gives out the PC private IP-Adresses of Int1/2 Asterisk can't give out a public IP-address for Int1/2. Where would it get one from? Correct that it doesn't. But some kind sould could indeed code a variety of techniques to get it, such as: a) If the phone supports STUN or a static external IP mode, it often will include the known external IP in its SIP headers. Asterisk would ignore those headers and talk directly to the internal address (equivalent to what it does with NAT=yes, but for RTP too) when talking directly to the phone, but when connecting the phone to outside addresses, it would pass these addresses and SDPs. b) In many cases, such phones sit behind a NAT which can have hard coded port forwardings so that Asterisk could simply be told, in sip.conf, what the external IP and port of the phone are, and it could rewrite SDPs when needed. c) The phone could attempt to reach Asterisk through an external address, even though the phone is on the same natwork as the Asterisk. This would tell Asterisk the external IP of the phone, and it could rewrite SDPs. However, since many NATs don't support hairpin of audio, for those NATs asterisk must use internal SDPs to connect devices behind the same NAT, including itself if that is the case. Anyway, as noted, there are many answers to your question of how asterisk _could_ get the info from, but at present it does not. They are all kludges of course. Just about all NAT penetration involves some kludging. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
Brad Templeton schrieb: On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote: PC When I set for Extern1/2 canreinvite=yes it works, but PC Intern-2-Extern doesn't work because Asteisk gives out the PC private IP-Adresses of Int1/2 Asterisk can't give out a public IP-address for Int1/2. Where would it get one from? Correct that it doesn't. But some kind sould could indeed code a variety of techniques to get it, such as: Again: My Problem is not Intern-to-Extern (NAT,Stun). My Problem is Extern-to-Extern, that the external phones are not talking RTP *directly* to each other. This is bad, when Asterisk is in Europe and the Phones are in Asia. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
On Tue, Jan 30, 2007 at 12:00:17PM +0100, Patrick Cervicek wrote: Brad Templeton schrieb: On Mon, Jan 29, 2007 at 12:05:28PM +0100, Benny Amorsen wrote: PC When I set for Extern1/2 canreinvite=yes it works, but PC Intern-2-Extern doesn't work because Asteisk gives out the PC private IP-Adresses of Int1/2 Asterisk can't give out a public IP-address for Int1/2. Where would it get one from? Correct that it doesn't. But some kind sould could indeed code a variety of techniques to get it, such as: Again: My Problem is not Intern-to-Extern (NAT,Stun). My Problem is Extern-to-Extern, that the external phones are not talking RTP *directly* to each other. This is bad, when Asterisk is in Europe and the Phones are in Asia. ___ That you can usually make work with most phones today, as long as the phone has some NAT penetration in it.The most versatile approach is STUN -- see if you can configure STUN on your phones. Some phones that don't have STUN do support a hard-configured external address. However, they often require a numeric IP, which means that they only work if you have a static IP, and your RTP port number never changes on the outside. A few phones also support extracting information from rport fields etc. However, they tend to have STUN too. Then you must also keep a port in the NAT open, by one of the following methods. a) Manual hole in the NAT -- most NATs support this (port forwarding) and it is recommended. Set your phone to a fixed SIP Port and port forward that to your phone. Requires phone have a fixed internal IP usually. b) Keep alive transmitted by phone (will be on phone's config) c) qualify=yes in asterisk sip.conf sends keep alive from asterisk. Works, but if it ever fails your phone won't ring until it re-registers. d) Very short registration time, like 30 seconds to 1 minute. This will effectively do a keep alive from the phone. I like this least. If your phone does not understand NAT properly it will forward a useless SDP that is internal. This is what some Cisco phones do. For those, you are currently screwed. Some day Asterisk might support rewriting SDPs that contain unreachable addresses but it does not, at present, do this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: NAT: RTP Path Optimization
PC == Patrick Cervicek [EMAIL PROTECTED] writes: PC But then all RTP Traffic of my internal phones will go over PC Asterisk. I want RTP to go Peer-to-Peer. == Intern-2-Intern PC and Extern-to-Extern should go P2P and Intern-2-Extern should PC go over Asterisk, see picture I understand what you want. I am telling you that you cannot get what you want, and the best compromise you can achieve. Either your internal-to-internal calls go direct, or your external-to-external do. Pick one. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
On Tue, Jan 30, 2007 at 10:23:09PM +0100, Benny Amorsen wrote: PC == Patrick Cervicek [EMAIL PROTECTED] writes: PC But then all RTP Traffic of my internal phones will go over PC Asterisk. I want RTP to go Peer-to-Peer. == Intern-2-Intern PC and Extern-to-Extern should go P2P and Intern-2-Extern should PC go over Asterisk, see picture I understand what you want. I am telling you that you cannot get what you want, and the best compromise you can achieve. Either your internal-to-internal calls go direct, or your external-to-external do. Pick one. You can almost get it, if your NAT will hairpin, by having your internal phones all present their external addresses. Then all phones will appear external, and all can talk to one another (though there can still be port change problems on reinvite if you don't do explicit ports) -- but internal to internal (and all other internal involved calls) will go through your NAT box, not your Asterisk box. However, the NAT must hairpin audio, and a lot of them don't. Here is Cullen's latest chart of who hairpins: http://tools.ietf.org/wg/behave/draft-jennings-behave-test-results-03.txt Only a few do it, and only a few make his group A category at all. Of course, many of these boxes are not expensive so it may be worth switching, though many of us love the WRT54G and its clones because of the ability to use open source firmware. However, the bastard does not hairpin, I don't know if any of the new firmwares do. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: NAT: RTP Path Optimization
PC == Patrick Cervicek [EMAIL PROTECTED] writes: PC http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is PC working fine in my Setup, but I want Extern1 to talk to Extern2 PC directly whitout going over Asterisk as the uplink is slow. PC When I set for Extern1/2 canreinvite=yes it works, but PC Intern-2-Extern doesn't work because Asteisk gives out the PC private IP-Adresses of Int1/2 Asterisk can't give out a public IP-address for Int1/2. Where would it get one from? PC I defined localnet=10.0.0.0/255.0.0.0 (Private LAN) but this PC doesn't help. PC Ideas, how to handle Extern-2-Extern (RTP bypass Asterisk)? Do I PC have to adjust nat somwhere? Set canreinvite to no for Int1/2. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: NAT: RTP Path Optimization
Benny Amorsen schrieb: PC == Patrick Cervicek [EMAIL PROTECTED] writes: PC http://lisas.de/~patrick/temp/rtp-optimierung.png Everything is PC working fine in my Setup, but I want Extern1 to talk to Extern2 PC directly whitout going over Asterisk as the uplink is slow. PC When I set for Extern1/2 canreinvite=yes it works, but PC Intern-2-Extern doesn't work because Asteisk gives out the PC private IP-Adresses of Int1/2 Asterisk can't give out a public IP-address for Int1/2. Where would it get one from? Asterisk gives out his own public adress and stays in the Media-Path between internal and external Phones. This avoids NAT-Problems. PC I defined localnet=10.0.0.0/255.0.0.0 (Private LAN) but this PC doesn't help. PC Ideas, how to handle Extern-2-Extern (RTP bypass Asterisk)? Do I PC have to adjust nat somwhere? Set canreinvite to no for Int1/2. But then all RTP Traffic of my internal phones will go over Asterisk. I want RTP to go Peer-to-Peer. == Intern-2-Intern and Extern-to-Extern should go P2P and Intern-2-Extern should go over Asterisk, see picture ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
