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Old 03-19-2007, 12:56 PM   #1 (permalink)
v164
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Default Which VoXaLot services are B2BUA?

Of the various VoXaLot services, my understanding is they can be divided into two groups, which I'll call "Group 1" and "Group 2":

Group 1:
VoXaLot acts as a SIP proxy server, just handling call signalling (SIP message packets). Once the call is established, voice traffic bypasses VoXaLot, unless VoXaLot determines the need for "NAT assistance" or transcoding.

Group 2 :
VoXaLot acts as a back-to-back User Agent (B2BUA). For these services the voice traffic (media) always passes through VoXaLot.


For "Group 1" services, voice traffic can bypass VoXaLot (for generally improved latency and quality) provided that the SIP devices are correctly configured (use of STUN, etc, if behind NAT, and supporting the required codecs).

However, for "Group 2" services, no matter how you configure your SIP device(s), the voice traffic will always transit VoXaLot.


So what I want to confirm is which VoXaLot services are in which group.

This is my understanding so far:

Group 1:
-Outbound SIP calls via VoXaLot (Dial plans, speed dial, ENUM/sip-code)
-Inbound SIP calls via VoXaLot
-Call forwarding to URI


Group 2: (implemented as B2BUA)
-Web Callback
-VoXCallMe
-Call forwarding to phone number via provider / dial plan


unsure:
-Provider Registrations (Inbound calls from a registered provider)
(I thought it would be "Group 1", but my tests seem to show the voice traffic transiting VoXaLot, although I'm not sure if it's because of codec negotiation or not).
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Old 03-23-2007, 10:28 PM   #2 (permalink)
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I thought this was a good question. Anyone of the admins able to verify or clarify this?
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Old 03-23-2007, 10:41 PM   #3 (permalink)
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Nice informative post e164. Have some reputation points from me

Answers in bold.

Group 1:
-Outbound SIP calls via VoXaLot (Dial plans, speed dial, ENUM/sip-code) Correct
-Inbound SIP calls via VoXaLot Correct
-Call forwarding to URI Correct


Group 2: (implemented as B2BUA)
-Web Callback Correct
-VoXCallMe Correct
-Call forwarding to phone number via provider / dial plan Correct


unsure:
-Provider Registrations (Inbound calls from a registered provider)
(I thought it would be "Group 1", but my tests seem to show the voice traffic transiting VoXaLot, although I'm not sure if it's because of codec negotiation or not).

For the most part these types of calls will end up as group 2. This is usually due to Codec negotiation resulting in VoXaLot transcoding or because the destination UAC is not initiating a re-invite.
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Old 06-23-2007, 08:48 AM   #4 (permalink)
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Quote:
Originally Posted by martin View Post

Group 2: (implemented as B2BUA)
-Web Callback Correct
-VoXCallMe Correct
-Call forwarding to phone number via provider / dial plan Correct

A possible exception, in the case of "Call forwarding to phone number via provider / dial plan" could be if, in the dial plan, there is an ENUM match, and the call gets sent to a SIP URI instead. Would that be "Group 1" in that case?
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Old 07-23-2007, 01:28 PM   #5 (permalink)
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How do you know if Voxalot is passing on your voice packets, or if you are working directly with the VSP?

I have two setups, one a Grandstream 496 ATA, the other an SJPhone softphone. I use voipcheap.com for outbound calls.

I would ideally like my voice packets to pass straight to voipcheap to reduce latency, but have no idea of checking whether they currently are or are not.
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Old 07-23-2007, 01:32 PM   #6 (permalink)
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Quote:
Originally Posted by The King View Post
How do you know if Voxalot is passing on your voice packets, or if you are working directly with the VSP?
First follow the steps in this tutorial:

HowTo: 6 Steps To Optimize Your Audio

The most definitive way is to check it yourself using something like Ethereal: A Network Protocol Analyzer
.
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Old 07-23-2007, 01:48 PM   #7 (permalink)
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Quote:
Originally Posted by Admin View Post
First follow the steps in this tutorial:

HowTo: 6 Steps To Optimize Your Audio

The most definitive way is to check it yourself using something like Ethereal: A Network Protocol Analyzer
.

That's what I'm using, Ethereal (or Wireshark: The World's Most Popular Network Protocol Analyzer as it's become).

Some SIP devices have a diagnostic log, or let you view status details, so you can check it there.
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