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-   -   Provider Registrations and Call Forwarding - is VoXaLot support really needed? (http://forum.sipbroker.com/showthread.php?t=1798)

v164 07-07-2007 01:46 AM

Provider Registrations and Call Forwarding - is VoXaLot support really needed?
 
There has been some discussion in this forum about Provider Registrations, and Call Forwarding - features which will not be available on VoXaLot basic. However, as long as the forwarding destinations are SIP URIs (ie, not phone numbers via VSPs, where credentials are required), it would seem to me that effectively equivalent functionality would be achievable even without VoXaLot's "Provider Registrations" and "Call Forwarding" features.

One of the things about RFC 3261 SIP, which I didn't realize until the MyNetFone + VoXaLot registrations issue earlier this year (see MyNetFone and Voxalot - MyNetFone - Voice over IP ) is that, in a SIP "REGISTER" message, the SIP URI in the "Contact: " header field does not have to be of the form "username@current_ip_address:listening_port", but can, it seems, be ANY valid SIP URI that resolves to where you want to be contacted for incoming calls (assuming the SIP service is RFC 3261-compliant in this regards).

The above link shows these two examples of SIP "REGISTER" messages:

Softphone registering with MyNetFone direct:
Code:

REGISTER sip:sip00.mynetphone.com.au SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;rport;branch=z9hG4bK151
From: MyNetPhone <sip:091xxxxx@sip00.mynetphone.com.au>
To: MyNetPhone <sip:091xxxxx@sip00.mynetphone.com.au>
Contact: "MyNetPhone" <sip:091xxxxx@xxx.xxx.xxx.xxx:5060>
Call-ID: 4A566BD314EF4FFB9C934023563286F7@sip00.mynetphone.com.au
CSeq: 28228 REGISTER
Expires: 3600
Max-Forwards: 70
Content-Length: 0

The SIP Softphone has created the Contact SIP URI as "<MNF account No.>@<current ip address>:<listening port>", a format that would be used 99% of the time, where a SIP device is registering with a SIP server.


VoXaLot registering with MyNetFone:
Code:

REGISTER sip:sip00.mynetfone.com.au SIP/2.0
Via: SIP/2.0/UDP 202.60.75.46:5061;branch=z9hG4bK15cc70ac;rport
From: <sip:091xxxxx@sip00.mynetfone.com.au>;tag=as3999d5b4
To: <sip:091xxxxx@sip00.mynetfone.com.au>
Call-ID: 521c0ce579d01b1422d9aa6908f8b13a@202.60.75.46
CSeq: 102 REGISTER
User-Agent: VoXaLot
Max-Forwards: 70
Expires: 3600
Contact: <sip:6xxxxx@202.60.75.46:5061>
Event: registration
Content-Length: 0

In this case, VoXaLot has created the Contact SIP URI as "<VoXaLot account No.>@<ip address of au.voxalot.com>:5061", a slight variation to the above case, but still RFC 3261-compliant.

(It appears that the MNF problem was that the above "99%" case was being assumed to apply 100% of the time, and as a result, the software was non-RFC 3261 compliant. I understand that MNF have since solved that particular problem.)


Anyway, VoXaLot accepts incoming SIP calls from anywhere (ie, VoXaLot, unlike some SIP services, "doesn't block incoming SIP calls from non-business-partner external domains"), and so it seems that all I would need to do to get incoming MNF calls to go to VoXaLot would be to use or create a SIP client to send a SIP REGISTER message to MNF (with my MNF credentials), and quote "<voxalot account No.>@voxalot.com" as the SIP URI in the "Contact: " header.

(The only difference I can see is that, when VoXaLot registers, the calls go to VoXaLot on port 5061, whereas normal SIP calls would go to VoXaLot on port 5060. I'm not sure what the difference is, although my earlier tests seem to show port 5061 as a kind of B2BUA, always proxying the voice stream (possibly for transcoding, etc), which in most cases I wouldn't mind not having.)


In the case of "Call Forwarding", that would be a simple matter of sending a similar SIP REGISTER message to VoXaLot, quoting in the "Contact: " header the SIP URI to which I want incoming calls forwarded. VoXaLot is known to support multiple registrations, so this would effectively provide "forking" for incoming calls (which, I understand, is not (yet) provided in the "official" VoXaLot Call Forwarding service).


So it seems that, if all you need is simple routing of SIP calls, using SIP URIs (not through any VSPs requiring credentials, and no advanced Call Forwarding options (time of day, diversion based on calling number, etc)), then whether or not your VoXaLot service supports "Provider Registrations" and "Call Forwarding" may not be relevant.

martin 07-07-2007 01:54 AM

0Yes your right v164, which is why VoxFree in our opinion is still a viable option for many people with all the core features still available.

Well spotted.
.

emoci 07-07-2007 05:25 AM

The problem is VoxFree, or VoxBasic, does not allow forwarding to a SIP URI.

This seems like something that could be included without harm or cost. Simply allow, that forwarding via SipBroker, or a plain SIP URI address be possible in the free plan, but you'd have to upgrade to use forwarding to a specific number, via a selected provider.

If you consider this in terms of traffic, this would create no more traffic than someone calling via VoxFree using their least cost routing.

Frankly most users (including myself) would have no idea how to prepare, and let alone sent a SIP message to the VoXalot servers as described above.


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