Latency explained?
 

When and what latency matters for calling overseas destinations?

Why does latency-to-SIP-server matter if packets go all the way between continents anyways?

Latency
 

SIP, when used in VoIP applications, is mostly a P2P protocol, so usually audio goes directly from source to destination, not through SIP proxy, which is used only to initiate the multimedia session. In this scenario, if you are making an overseas call, and the SIP server you are using is on the other side of the globe, your audio will not suffer from high latency because the two calling points are not so very far away from each other. The problem is when there is a need to use an audio proxy for transcoding or solving NAT issues. In this last scenario you should consider the latency introduced by the proxy being used. Some providers acts as audio proxying services, some don't. There is no universal rule, so you should understand the problem and have the possibility to know which are the technical specifications of the service being used. However, this is an information that most commercial departments can't give you, so you should discover it for your self through some testing of the provider.
Regards.

So in the case of Voxalot, does it act only to initiate the multimedia session and and does no proxying? What happens when calls are passed through another provider via Voxalot, such is Betamax. Do they proxy? Are there certain conditions when Voxalot will proxy and time when they wouldn't? Same question for the Betamax end.

Richard

Yes, if you have configured Voxalot to use re-invite. This happens if you have setted to 'yes' the 'Optimize Audio Path' in the Voice Service Provider configuration and 'Symmetric NAT Handling' to 'No' on the Member Details page. If you answer to the first option 'No' and to the second one 'Yes' (the defaults Voxalot configuration), then Voxalot will act as well as an audio proxy.

In theory, you may have a situation where Voxalot doesn't proxy your audio stream but your provider does it. This should happen if you dial a PSTN number, unless ENUM is used. You should try the best option that works for you. If you are at trouble, you can always use the Voxalot default configuration, meaning that Voxalot will proxy your audio stream as stated above. This should always work (no one-way-audio problem).
Regarding Betamax, I don't know because I've never used it. However, for the latency it seams to have (I know who use Voipbuster), it may be doing some audio proxying, but I can not confirm you that assumption.

Normally, no. However, if you have configured the audio optimization and the voice provider you are using doesn't accept re-invite, you may obtain only one-way-audio problem. In such circumstance, you should use the default configuration stated above.

I don't know as far as Betamax is concerned because I've never used this provider.

Regards.

Based on the results of my packet tracing tests done last year I can add a few details to the excellent information provided by carlosalbffgomes.

Even when 'Optimize Audio Path' is set to Yes, Voxalot will still proxy the RTP (voice) packets if it detects some kind of misconfiguration in the NAT settings of the party receiving the call, and possibly also the party making the call. This can result in unwanted, unexpected proxying.

Betamax will proxy all RTP packets for all calls and will not change RTP packet connection IP address after the call has been initiated. When placing a Betamax call through Voxalot this results in RTP packets making a big triangle path, with one side direct from the Betamax proxy (in The Netherlands) and the other side is a two-leg hop to The Netherlands via Voxalot's proxy.