On 2014-05-27, Andre Ruppert <a...@in-telegence.net> wrote: > You have two different protocols: SIP for signaling und RTP for media. > > Media information between the endpoints is specified in SIP-SDP-packets > (session description protocol). > > SDP-packets contain the original IPs of the VoIP-endpoints, and these > IPs won't be NATed!
It is common practice for voip providers (or, at least, voip providers who expect direct registrations from phones) to use the addresses from the packet headers, not from SDP. Either done directly in their SIP servers, or fixed up by their SBCs. > Do you make use of an sip-proxy or an external STUN-server at least? These aren't usually needed for "normal" voip providers. Wholesale providers may need the SDP addresses to be correct but you'll normally be using those with your own voip server and these usually just have static config to set the external address rather than using STUN.
