Best Quality - Sip Broker or VoXaLot?
 

Which peering service should I use, VoXaLot or SIP Broker.

I have set up direct IP dialling to my SPA3000 with a SIP Broker Alias and VoXaLot forwarding pointing to my DNS. Now that VoXaLot has upgraded its server, should I get my friends to call me on my VoXaLot number *01051xxxx rather than my SIP Broker number *0116151xx? ie will I get the benefits of less latency with the new VoXaLot server? I have my family in Adelaide calling the PSTN local number in Adelaide, and I have a choice of giving them the VoXaLot or SIP Broker number. FYI, the SIP Broker Ping is 231 ms average and the new VoXaLot is 30 ms

Matt

I think that you have answered your own question. Although once the call is setup I think Voxalot is out of the picture. There is a thread on this issue at WP.

Martin has answered several questions relating to this topic.
VoXaLot - ping times

I have read this thread, and am very interested in HOW to set up my ATA so it correctly routes the call bypassing VoXaLot once the SIP session has been initiated. This question was asked several times in the thread posted above, but has not been answered as yet. I assume the SIP message takes one path (via VoXaLot) and the UDP voice message takes a more direct path. I have not been able to experience this, so I assume my ATA is not set up as well as it could be.

Can someone help? Martin?

Regards

Matt

STUN, I think
 

If your ATA is behind a NAT (ie, it is not sitting on a public IP address), you need some way for the ATA to find out what the public IP address is, so it can put that address in the outgoing SIP messages.

One way is to use STUN, the other way would be to manually input the IP address. In addition, you would need to port forward some UDP ports for the incoming RTP stream (as well as UDP port 5060 for the incoming SIP INVITES).

Edited: more detail -

Basically, you need to get your ATA to send SIP INVITE messages out as if it was sitting on your public IP address.

Then, so you can actually receive the incoming audio stream from the other party when they answer, you need to either set port forwarding in your router, or use STUN to "punch a hole" for incoming RTP stream to get through - I think that's how STUN works.

Thanks v164, it helps with the concept of how it works. I'd like to find something showing enough detail to configure it. We might learn more about SIP invites and responses along the way.

Anyone have any good links on this?

If you're using Linksys or Sipura try here.

The recommended settings found at this link work great on my adapter (SPA-2100) and allow it to 'punch holes' in the router's firewall so that I can use STUN entirely to identify my public IP address and to open up the necessary TCP and UDP ports for me so that my ATA requires zero configuration on a non-symmetric NAT router (which is most of them). My only difference in settings from the ones at this wiki are that my "STUN Test Enable" is set to yes, and "STUN Server" is stun.xten.com (since I found FWD to go down entirely from time to time--which would kill my VOIP for as long as the stun server was down).

FYI, VoXaLot have a stun server also:

stun.voxalot.com.au:3478

Any explanation as to why your Stun test is enabled but they recommend to disable?

STUN test is a setting that when added to the other ones, checks to see if your connection really needs NAT mapping and other NAT settings enabled or disabled and depending on what it discovers will modify your session settings accordingly, even if user specified otherwise, to ensure a more seamless functioning.