When do RTP packets go via proxy server instead of direct?
 

When calling from one Voxalot registered SIP phone to another, I noticed that sometimes the RTP (voice) packets travel through Voxalot's proxy, and sometimes by a direct route to the other party. I could not find any documentation about this issue, but it seems that if the receiving party's NAT settings are correct then the RTP packets go directly between the two endpoints. On the other hand, if the receiving party's NAT settings are incorrect then the RTP packets will be sent through Voxalot's proxy for the entire call duration.

Does anyone know more about this?

How do you check if voice packets go directly or through Voxalot proxy?

I use Wireshark. The test procedure is simple; just call a Voxalot number and observe the Source and Destination IP addresses of the RTP (voice) packets. This issue is important because RTP packets sent through a proxy can be delayed or dropped, causing poor audio quality.

Many Voxalot users assume the RTP packets always take the most direct route through the Internet. This is not always the case, particularly when calling out through one of the Betamax companies, that's been my experience. I have also seen the Voxalot proxy handling RTP packets when I place a call to a Voxalot user who's ATA is behind a router and has mis-configured NAT settings.

Good observation. It is a common misconception that if the callers ATA is configured to handle NAT issues that the RTP streams will be point to point.

To enable the 2 RTP streams (both inbound and outbound) to be point to point requires both end points to be properly setup to handle NAT.

The Voxalot proxies are "smart" enough to proxy RTP if and when required.

Hi Martin,

I began investigating this issue in order to solve a problem described in this post.

Can you recommend a SIP number to call for testing NAT traversal settings? I have been telling people to use *0@proxy01.sipphone.com but it would be helpful to have more details than what that number gives.

STUN is bad idea?
 

Speaking about STUN as a condition for correct NAT traversal, take a look at this post:

[Groupe uk-telecom-voip] : Re: Newbie Question - about using a DECT SIP phone and port forwarding - Article

See this thread:

http://forum.voxalot.com/voxalot-sup...s-voxalot.html