Re: [asterisk-users] Psst... Top secret information: Codename Pineapple
14 okt 2006 kl. 09.44 skrev Brian Candler: On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote: * Phones = stations, regardless of where they are Asterisk = SIP Server, Phone = SIP Client * Trunks = trunks to other SIP servers, bilateral Asterisk and the other server is peer to peer * Services = services you register for, like BroadVoice, Voop or FWD. (where asterisk acts as a phone) Asterisk = SIP Client, Other End = SIP Server Hmm, but I don't see how these ideas map to formal SIP concepts (RFC 3261). Let's try to clarify then. phones are devices that connect to Asterisk. They register with Asterisk acting as a SIP location server/registrar and use Asterisk as the outbound SIP proxy. They get calls from Asterisk and place calls to Asterisk. The phone use one of the SIP domains that are hosted within your Asterisk server. (this is like the current friend) service is when Asterisk is the UA, acting as a phone towards another SIP server - we register with a SIP location server/registrar to get incoming calls. We place calls, masquerading as a phone (using the registrars domain). Currently, this is a mixture between a peer (matched on IP for incoming calls) and a register= statement. In some cases, two peers and a register= statement. Very confusing. trunk is when we exchange traffic with another server. We send calls to their SIP domain and receive calls to our SIP domain. We may use realm based authentication for the incoming part of the trunk (not based on caller ID/From: header) and a combination of SIP domain and ACLs. This is currently handled by defining sip peers for outbound calls and separate SIP peers for inbound calls - where we match on IP. The problem with the IP matching is when a trunking partner use several SIP servers to connect to us, we need to define one peer per server instead of just matching on domain and then authenticate. In all cases, we're a SIP user agent client/server in SIP terminology. In fact, we're a super-SIP ua called a B2BUA. I am trying to avoid sip client since the whole user/peer client/server concept does not really match SIP. In some cases, we're the SIP registrar/location server and in other we're configured as the outbound proxy, even though we are not a proxy. I hope I did not add to the confusion by this confusing message. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ - Stockholm, Sweden, November 13-17 ___ --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] Psst... Top secret information: Codename Pineapple
Brian Candler wrote: On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote: * Phones = stations, regardless of where they are Asterisk = SIP Server, Phone = SIP Client * Trunks = trunks to other SIP servers, bilateral Asterisk and the other server is peer to peer * Services = services you register for, like BroadVoice, Voop or FWD. (where asterisk acts as a phone) Asterisk = SIP Client, Other End = SIP Server Hmm, but I don't see how these ideas map to formal SIP concepts (RFC 3261). Phone = User Agent Client (places outgoing calls) and also User Agent Server (accepts incoming calls) But then Asterisk is both of these too. The term SIP Client does not appear in RFC 3261 at all. The term SIP Server does, in a loose generic way, when they mean SIP Proxy and/or SIP Registrar. Asterisk is never a SIP Proxy, it's a SIP endpoint (UAC/UAS). I think it *is* a registrar though. So what I'm asking is: what's fundamentally different between a phone, and trunk, and a service? How does Asterisk treat them differently? After all, placing a SIP call to a phone (via a dialplan) and routing a SIP call down a trunk (via a dialplan) are the same operation, aren't they These ideas don't map to formal SIP concepts. Olle's ideas seemt to map to more formal Asterisk concepts. My terms are more generic and try to map to layman's internet concepts. Really, a SIP device is a SIP device. All SIP devices are clients and all SIP devices are servers. It's how you USE the device. ___ --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] Psst... Top secret information: Codename Pineapple
* Phones = stations, regardless of where they areAsterisk = SIP Server, Phone = SIP Client Is a Media Server a Phone (ie SIP Client) ? ___ --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] Psst... Top secret information: Codename Pineapple
On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote: * Phones = stations, regardless of where they are Asterisk = SIP Server, Phone = SIP Client * Trunks = trunks to other SIP servers, bilateral Asterisk and the other server is peer to peer * Services = services you register for, like BroadVoice, Voop or FWD. (where asterisk acts as a phone) Asterisk = SIP Client, Other End = SIP Server Hmm, but I don't see how these ideas map to formal SIP concepts (RFC 3261). Phone = User Agent Client (places outgoing calls) and also User Agent Server (accepts incoming calls) But then Asterisk is both of these too. The term SIP Client does not appear in RFC 3261 at all. The term SIP Server does, in a loose generic way, when they mean SIP Proxy and/or SIP Registrar. Asterisk is never a SIP Proxy, it's a SIP endpoint (UAC/UAS). I think it *is* a registrar though. So what I'm asking is: what's fundamentally different between a phone, and trunk, and a service? How does Asterisk treat them differently? After all, placing a SIP call to a phone (via a dialplan) and routing a SIP call down a trunk (via a dialplan) are the same operation, aren't they? Maybe we need to authenticate to the other side. Maybe we want to require the other side to authenticate to us. But AFAICS that's something you might want to do set (or not) for any SIP endpoint. For instance, you might want to say that all devices with IP address 192.168.1.x can place calls without authentication. Regards, Brian. ___ --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] Psst... Top secret information: Codename Pineapple
On Thu, Oct 12, 2006 at 09:11:29PM +0200, Olle E Johansson wrote: [ quoting me: ] Does that mean that it will make a distinction concerning the difference in administrative span of control between trunks, which go to the outside world, and stations, which are part of your PBX (even though they may *be* out in the world somewhere, anyway? Right. To explain a bit further: * Phones = stations, regardless of where they are * Trunks = trunks to other SIP servers, bilateral * Services = services you register for, like BroadVoice, Voop or FWD. (where asterisk acts as a phone) I would suggest that anything that carries incoming or outgoing calls from the administrative span of control of your * server to somewhere else ought to be a trunk; IE: I'm not sure what distinction you're making between items 2 and 3. Could you clarify? Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] Psst... Top secret information: Codename Pineapple
On Oct 13, 2006, at 6:12 PM, Jay R. Ashworth wrote: On Thu, Oct 12, 2006 at 09:11:29PM +0200, Olle E Johansson wrote: [ quoting me: ] Does that mean that it will make a distinction concerning the difference in administrative span of control between trunks, which go to the outside world, and stations, which are part of your PBX (even though they may *be* out in the world somewhere, anyway? Right. To explain a bit further: * Phones = stations, regardless of where they are * Trunks = trunks to other SIP servers, bilateral * Services = services you register for, like BroadVoice, Voop or FWD. (where asterisk acts as a phone) I would suggest that anything that carries incoming or outgoing calls from the administrative span of control of your * server to somewhere else ought to be a trunk; IE: I'm not sure what distinction you're making between items 2 and 3. Could you clarify? I have the same trouble understanding the difference between 2 and 3. --- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? PGP.sig Description: This is a digitally signed message part ___ --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] Psst... Top secret information: Codename Pineapple
* Phones = stations, regardless of where they are Asterisk = SIP Server, Phone = SIP Client * Trunks = trunks to other SIP servers, bilateral Asterisk and the other server is peer to peer * Services = services you register for, like BroadVoice, Voop or FWD. (where asterisk acts as a phone) Asterisk = SIP Client, Other End = SIP Server Could you clarify? I have the same trouble understanding the difference between 2 and 3. ___ --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] Psst... Top secret information: Codename Pineapple
On Wed, Oct 11, 2006 at 09:00:20AM +0200, Olle E Johansson wrote: The new channel will have configurations for trunks, services and phones. It will Does that mean that it will make a distinction concerning the difference in administrative span of control between trunks, which go to the outside world, and stations, which are part of your PBX (even though they may *be* out in the world somewhere, anyway? That's a spot that my (admittedly loose) understanding of SIP has always sort of glossed over... Cheers, -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --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] Psst... Top secret information: Codename Pineapple
12 okt 2006 kl. 03.36 skrev Andrew Joakimsen: What are your T.38 plans with this? That's top secret... :-) The T38 will be handled the same way as today - in passthrough mode - until we have more T38 implementation code within the core. That's a bit outside of the SIP scope. /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] Psst... Top secret information: Codename Pineapple
12 okt 2006 kl. 15.51 skrev Jay R. Ashworth: On Wed, Oct 11, 2006 at 09:00:20AM +0200, Olle E Johansson wrote: The new channel will have configurations for trunks, services and phones. It will Does that mean that it will make a distinction concerning the difference in administrative span of control between trunks, which go to the outside world, and stations, which are part of your PBX (even though they may *be* out in the world somewhere, anyway? Right. To explain a bit further: * Phones = stations, regardless of where they are * Trunks = trunks to other SIP servers, bilateral * Services = services you register for, like BroadVoice, Voop or FWD. (where asterisk acts as a phone) Regards, /Olle ___ --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] Psst... Top secret information: Codename Pineapple
What are your T.38 plans with this? On 10/11/06, Olle E Johansson [EMAIL PROTECTED] wrote: Friends in the Asterisk community, I've been talking for years about the new version of the SIP channel. I've been trying to get funding and get going. Well, the funding part remains to be handled, but I have other news - if you kan keep it to yourself. ...I've began coding. Finally. With a happy smile on my face I removed pedantic=yes the other day. After years of disliking that option it's gone! And srvlookup now defaults to yes in the source code :-) So what is the chan_sip3 project (codename pineapple) about? -- The current SIP channel has many code relationships to the IAX2 channel. Concepts like users, peers and friends doesn't really fit the SIP architecture. The channel supports locally connected phones very well, but is having severe problems being part of a larger SIP infrastructure. Forking, branching and such is not handled, as well as multiple transactions at the same time. The new channel will have configurations for trunks, services and phones. It will be more domain-focused to support multihosting better. It will have a proper SIP state machine so we can handle TCP and TLS alongside UDP. It will have STUN support, like the current Google talk channel. And a lot of other changes... Can I test this now? -- Don't expect this work to be completed yesterday. Right now, I'm cleaning up stuff, moving around variables, splitting up the code in multiple files and grouping variables into structures. When all of that is done, the real work will start. I am expecting to have an experimental version ready for the release of Asterisk *after* the 1.4 release and a more production-ready version ready for the release a year from now. As always with Open Source, the final result depends a lot on the help from the community in testing, providing fixes, development time, funding and additions. Is it available for download? --- The code is hosted in the codename-pineapple branch in the svn server. In that branch, there's a chan_sip.c (version 1) and a chan_sip3.c. As I said: don't expect much yet and don't run this in production! Right now, downloading it is a good way of wasting the bytes on your hard disk drive and not much more. In Q1 2007 I will run an AstriSIPcon developer's meeting to be able to meet everyone that has interest in Asterisk and SIP to test, discuss and work with the new SIP channel. SIP greetings! /Olle PS. A big thank you to Voop AS, who keeps supporting my development work with Asterisk as well as all the students in my training classes that provide development funding by attending the classes. Thanks! --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Next class: Stockholm, Sweden November 13-17 2006 ___ --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 ___ --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
