Re: Multiple SIP Clients?

Anthony R. Gold wrote:
> Can one have a SIP incoming call ring in multiple places and over multiple
> ATAs and softphones such that the first one to pick-up will seize the call?

Yes, that is a standard part of SIP, and implementable at the proxy
level; it is called branching. It may not be a service operated by
public PSTN gateways, though.

> My present configurations have each client registering with the server and so
> causing a pre-existing registration to be closed and ended.

I'm not sure that multiple registrations against the same address of
record is allowed though. Asterisk certainly doesn't support it. You
would probably have to make to do some hard configuration in the proxy
or B2BUA. For Asterisk (a B2BUA) you put
......Dial(SIP/address1&SIP/address2&SIP/address3&otherprotocol/address4)
in extensions.conf, or the "dialplan" database.

One key to this is maintaining the distinction between directory numbers
(Asterisk extensions) and equipment numbers (Asterisk devices), which is
often confused when the switching equipment allows similar names for both.

"David Woolley" <david@ex.djwhome.demon.invalid> wrote in message
news:iu4ccp$r5j$1@dont-email.me...
> Anthony R. Gold wrote:
>> Can one have a SIP incoming call ring in multiple places and over
>> multiple
>> ATAs and softphones such that the first one to pick-up will seize the
>> call?
>
> Yes, that is a standard part of SIP, and implementable at the proxy level;
> it is called branching. It may not be a service operated by public PSTN
> gateways, though.

Voipfone claim to be working on it, but not sorted yet AFAIK.

>
>> My present configurations have each client registering with the server
>> and so
>> causing a pre-existing registration to be closed and ended.
>
> I'm not sure that multiple registrations against the same address of
> record is allowed though. Asterisk certainly doesn't support it. You
> would probably have to make to do some hard configuration in the proxy or
> B2BUA. For Asterisk (a B2BUA) you put
> .....Dial(SIP/address1&SIP/address2&SIP/address3&otherprotocol/address4)
> in extensions.conf, or the "dialplan" database.
>
> One key to this is maintaining the distinction between directory numbers
> (Asterisk extensions) and equipment numbers (Asterisk devices), which is
> often confused when the switching equipment allows similar names for both.

In article <6c6f07p6g03574vltab2u6kisg6rs2i2q8@4ax.com>,
Anthony R. Gold <tgold@panix.com> wrote:

>New question: is there any limit to the number of SIP devices or softphones
>that may be connected to one home LAN? Is it useful for different ones be set
>to different UDP port numbers? And if so, what is the range of port numbers
>that are generally accepted?

Within reason, no... However who knows what a reasonable limit is.

Factors that will affect it include the size of the ARP table in your
router and the MAC to port addressing tables in your Ethernet switches.
Also your routers ability to manage NAT sessions too. (And routers with
broken SIP ALGs that are really only expecting one SIP device)

Then there's the DHCP server and your LAN subnet - once you get over
254 devices on one LAN you need a bigger subnet.

Your router doing NAT should give each device it's own outgoing port
number - which is one way NAT uses to map incoming connections to
a device, so you really shouldn't have to fiddle with SIP phones at
that level. (And if you don't have NAT, then there's even less issues)

So basically it's your router and to a lesser extent Ethernet switches
that's going to be the limiting factor.

Gordon

"Anthony R. Gold" <not-for-mail@ahjg.co.uk> wrote in message
news:6c6f07p6g03574vltab2u6kisg6rs2i2q8@4ax.com...
> On Sun, 26 Jun 2011 19:06:51 +0100, "R. Mark Clayton"
> <nospamclayton@btinternet.com> wrote:
>
>>
>> "David Woolley" <david@ex.djwhome.demon.invalid> wrote in message
>> news:iu4ccp$r5j$1@dont-email.me...
>>> Anthony R. Gold wrote:
>>>> Can one have a SIP incoming call ring in multiple places and over
>>>> multiple
>>>> ATAs and softphones such that the first one to pick-up will seize the
>>>> call?
>>>
>>> Yes, that is a standard part of SIP, and implementable at the proxy
>>> level;
>>> it is called branching. It may not be a service operated by public PSTN
>>> gateways, though.
>>
>> Voipfone claim to be working on it, but not sorted yet AFAIK.
>
> Does anyone know about VOIPCheap, with whom I was having that problem?
> After
> Paul mentioned his experience I was then successful with sipgate.
>
> New question: is there any limit to the number of SIP devices or
> softphones
> that may be connected to one home LAN? Is it useful for different ones be
> set
> to different UDP port numbers? And if so, what is the range of port
> numbers
> that are generally accepted?

It will depend on the speed of your uplink, but even a now modest 1Mb/s will
give you several simultaneous connections.

In article <iu4ccp$r5j$1@dont-email.me>,
David Woolley <david@ex.djwhome.demon.invalid> writes:
> Anthony R. Gold wrote:
>> Can one have a SIP incoming call ring in multiple places and over multiple
>> ATAs and softphones such that the first one to pick-up will seize the call?
>
> Yes, that is a standard part of SIP, and implementable at the proxy
> level; it is called branching. It may not be a service operated by
> public PSTN gateways, though.
>
>> My present configurations have each client registering with the server and so
>> causing a pre-existing registration to be closed and ended.
>
> I'm not sure that multiple registrations against the same address of
> record is allowed though. Asterisk certainly doesn't support it. You

Works fine with SER. Didn't have to do any special config.

--
Andrew Gabriel
[email address is not usable -- followup in the newsgroup]