You can make things even complicater - e.g. if you Answer() before Dialing out to the other client (e.g. for some announcements) - then directrtp wont work without reINVITE.
regards klaus Adam Gundy wrote: > following on from a suggestion that a bug report I raised (10335) is at > least partly a feature request and should be discussed on -dev, I have a > question about the meaning of 'nat=yes', and a feature request... > > basically, I have some NAT-blind SIP clients (OpenWengo) which do not > support SIP reinvites, and I was hoping the directrtpsetup option in > asterisk 1.4.x would help with this. > > the problem is that it doesn't work, basically because of the way > 'nat=yes' deals with RTP - asterisk waits for an incoming packet, then > sends all RTP to the IP/port that it came from. this works around the > fact that the NAT-blind SIP client put LAN IP/ports in the SIP packet. > > so, when we get to directrtpsetup=yes, asterisk ends up sending the LAN > IP/ports of the two clients to each other, which obviously doesn't work > (unless they both happen to be behind the same NAT!), because asterisk > *never receives any RTP from the clients to fix its idea of the IP/ports*. > > now here's the real question: what does 'nat=yes' mean? if it implies > 'symmetric NAT' or 'router with ports forwarded', then we can actually > fix up this situation; we DO know the IP address that the SIP packet > came from, and if (as in my case) the SIP IP address is the same as the > RTP IP address, we could fix our idea of the RTP address without any > packets arriving, and the direct connection should work. > > alternatively, if 'nat=yes' means 'dumb SIP client with no clue what its > IP/ports are', then we basically have to ban directrtp for this case, > and either use reinvites (if enabled), or proxy. > > if that IS the case, can we have a 'nat=symmetric' which means > 'symmetric NAT/router with ports forwarded' please? I'm sure there are > many SIP clients or hardware widgets out there which have no idea about > STUN and basically come with instructions to forward a range of ports > (I'm looking at the Polycom phone on my desk here as one example) > > thanks.. > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
