Your ATA/Sip Phone/Soft Phone determines what RTP ports will get used. You would forward the RTP ports that it will use to that particular device.
AFAIK, SIP only uses a single port. The default is 5060, but you can also use 5061, 5062, etc.
With multiple VoIP devices, you would have to set each device to a particular range of RTP ports and use multiple forwarding entries to route the appropriate ranges to the appropriate device. Skype doesn't use SIP/RTP and handles itself properly in most cases without any help.
The relationship between SIP and RTP is extremely unclean at the protocol design level when it comes to routing behind and through NAT routers. This has nothing to do with Voxalot. Audio problems with VoIP where NAT routers are involved are a significant problem with all providers and aggregators. I wish I could point you to a one-size fits all cookbook set of entries to make in your router, but it simply doesn't exist.
|