[Asterisk-Users] New Asterisk bounty: SIP simultaneous

[Olle E. Johansson]

Paul Mahler wrote:

> Well, this is certainly getting exciting. 
Yes, it is. Sorry for coming in late to this debate...

> Andy, I took your advice and re-read the RFP. 
It's actually RFC, not RFP. (teasing :-)

  > So, gentlemen, help me out here. The spec says:
> 
> "The Address of record is the "SIP address that the registry knows the
> registrand. .  .""

The Address of record is the public SIP uri you want people to call you at,
regardless of the address of the phone you are answering on. It's the
SIP phone address you place on your business card.

> "A client uses the REGISTER method to register the address listed in the TO
> header field with a SIP Server."

A client registers a temporary address, the address to a SIP UA, to the
SIP registrar that is responsible for the domain in the AOR. This tells
the SIP registrar (or "location server") where to find you if someone
calls your URI.

When sending mail, I am not addressing the mail to the IP address you
are reading the mail on, I am using your public e-mail address that is mapped
to an e-mail server that is responsible for all e-mail to your domain.
Later on, you fetch the e-mail from an e-mail client somewhere, with an
IP address that propably changes as you travel around signing books  ;-)

SIP works the same way. You have a public address and a SIP proxy being
responsible for keeping track of where you want to answer your calls.
You can surely register several phones that you want to answer on.
The proxy takes care of hiding this to the callee, so that the caller
only get one set of responses. That's what the "forking" stuff is all about.
If one phone is busy and the other one is answering, we should only signal
"200 OK" in SIP lingo to the caller.

> I don't see how two different clients can register with a server as the same
> address of record. Doesn't the second registration from a new client change
> the address of record for the registered client?

You have one address-of-record that maps into several SIP URIs, one for each
device. These are not as long-term as your a-o-r SIP URI.
 From the RFC:

"Location Service: A location service is used by a SIP redirect or proxy server
to obtain information about a callee’s possible location(s). It contains a
list of bindings of address-of-record keys to zero or more
contact addresses. The bindings can be created and removed in many ways;
this specification defines a REGISTER method that updates the bindings."

> If the second client is trying the same registration as the first client,
> and it's the responsibility of the client to provide the complete list of
> bindings, how does the second client know the list of bindings for the first
> client that bound the registration? 

It's *not* the responsibility of the *client* to provide a list, it's the
server that responds with a list, telling the client "by the way, these
devices are also registred for the same a-o-r."

> So isn't this the problem * has? The first client registers as the address
> of record, then the second client comes in with the same registration and
> becomes the address of record? 

The address of record does not change because of a registration. The stored
address (the contact: header) of where we can reach you (location) changes.
And yes,  if you have multiple devices registering for the same Asterisk sip [peer]
account, it will be changing for each registration. This is not the behaviour
of most SIP Proxys.

Asterisk is *not* a SIP proxy. It's a SIP registrar and location server.
It's a very clever SIP UA. It wants to be in the middle of the call
and wants to be in control of each device. This device-slave view doesn't
match the SIP architecture. Due to Asterisk's multi-protocol architecture
we have to make some compromises in the SIP channel to be able to have
some kind of generic view of calls and phones in the core.

A SIP proxy is never the end point of a call, it should never handle
the media stream. The power is in the edge, in the phones. This is why
transfers and other PBX functions is a bit messy with SIP and Asterisk,
we are trying to find a way to do it centralized as Asterisk but de-
centralized as SIP...

I've spent a considerable amount of time investigating support for multiple
registrations on one Asterisk sip [peer] account and after learning about
Asterisk's architecture come to the conclusion that it is not an easy or even a
desirable feature to implement. The architecture of Asterisk is a PBX, and the dial
plan and a lot of apps wants to be in control of the device.

It may be possible, but will probably lead to a lot of changes to Asterisk,
both core and applications, that no other channel will benefit from. A quick
hack to support it may lead to a lot of confusion on how to handle other apps.
And it's a lot more work than the bounty will cover. I suggest that you use a
forking SIP proxy in conjunction with Asterisk to get this functionality.

If you are looking for a SIP PBX, check Pingtel's Open Source software.
If you are looking for a SIP proxy, test SIP Express Router from iptel.org.
If you are looking for a brilliant multi-protocol open source PBX with SIP support,
stay with Asterisk :-)

Executive summary of a long posting:
* Yes, SIP supports multiple registrations on one SIP account
* No, I do not think Asterisk should support this with the current
   architecture and
* No, there's no quick fix for a 100 USD bounty

Asterisk is a very large telephony platform, covering many areas and protocols. Very
few people can know all the details of all the protocols and interfaces we support.
I don't know much either on H.323 or PRI signalling...

I don't think we should let these misunderstandings judge the quality of Paul's
Asterisk book. Even authors need to learn now and then :-)

/O