Re: 2 VoIP phones on one line
On 2014/10/24 11:00, Rod Whitworth wrote: None of the STUN etc that have arrived, since I last looked, any good? STUN can help with weak forms of NAT as used on some simpler routers but won't help with a symmetric NAT (i.e. one that checks both the port numbers and addresses, like PF - if you're only used to PF NAT then reading a description of full cone NAT might come as a surprise ;) For symmetric NAT, one way or another you need the call media to come from a known point that the side behind NAT communicates with. The simple way to do this is by ignoring the addresses in SIP/SDP headers, using the packet addresses instead, and always proxying call audio (media), replying to the same port that they see your media coming from. comedia mode. This is quite commonly used in voip providers (either on their SIP registration servers, or via separate SBCs - session border controller appliances). There are other ways to do it, which can reduce (or at least spread) resource use on the provider side in situations with weak NAT / fully open SIP devices, but they mean 1) several different methods used on the provider side used in different situations (harder to debug) and 2) relying on the client to correctly implement more modern protocols. If I was a big voip provider I would likely only want to use these methods if I controlled the client side software as well as the server. In any event they only optimize decreasingly common cases. It's a different case for wholesale providers rather than the type of provider you'd use directly for phones - in that situation, it's fairly important for them to be able to split media and signalling (both for flexibility and to reduce latency) but making it easy for clients behind NAT isn't really a concern. Even in those cases you can often still work behind NAT without any special protocols (most phones/ATAs let you set your known external IP to use in headers, send frequent keepalives, and choose a particular port range for RTP - then port forwarding takes care of the rest). Summary: try it the easy way first - ignore all this stuff, just put the phone behind NAT without anything special, and see if it works. If it doesn't then try plan B. If it still doesn't work, you can get clues by watching pflog and the SIP messages (it's a text based protocol; tcpdump -s 1500 -A port 5060 will be readable, though ngrep -d iface -W byline port 5060 is probably easier on the eye).
2 VoIP phones on one line
Years ago I bought a Minitar VoIP ATA and it was great. Then SWMBO wanted one too and I set up siproxd which mostly worked. Then I go two global IPs and put one ATA/Phone on each. Perfect! No Siproxd! Now I am about to need those 2 IPs. Neither phone needs to recieve incoming calls. Anybody using some of the more recent additions inthe Telephony ports? Ones to avoid or ones to love? Thanx, *** NOTE *** Please DO NOT CC me. I am subscribed to the list. Mail to the sender address that does not originate at the list server is tarpitted. The reply-to: address is provided for those who feel compelled to reply off list. Thankyou. Rod/ --- This life is not the real thing. It is not even in Beta. If it was, then OpenBSD would already have a man page for it.
Re: 2 VoIP phones on one line
On 2014-10-23, Rod Whitworth glis...@witworx.com wrote: Years ago I bought a Minitar VoIP ATA and it was great. Then SWMBO wanted one too and I set up siproxd which mostly worked. Then I go two global IPs and put one ATA/Phone on each. Perfect! No Siproxd! Now I am about to need those 2 IPs. Neither phone needs to recieve incoming calls. Anybody using some of the more recent additions inthe Telephony ports? Ones to avoid or ones to love? Thanx, I would first try it with standard NAT. It depends on the exact setup on both sides, but it's common for VoIP providers these days to handle various NAT based configurations in their border controllers or SIP servers - the first troubleshooting step from their side would often be to *disable* any nat helpers. If that's no good, try restricting each ATA/phone to a different RTP port range and port-forward them.
Re: 2 VoIP phones on one line
On Thu, 23 Oct 2014 21:24:56 + (UTC), Stuart Henderson wrote: On 2014-10-23, Rod Whitworth glis...@witworx.com wrote: Years ago I bought a Minitar VoIP ATA and it was great. Then SWMBO wanted one too and I set up siproxd which mostly worked. Then I go two global IPs and put one ATA/Phone on each. Perfect! No Siproxd! Now I am about to need those 2 IPs. Neither phone needs to recieve incoming calls. Anybody using some of the more recent additions inthe Telephony ports? Ones to avoid or ones to love? Thanx, I would first try it with standard NAT. It depends on the exact setup on both sides, but it's common for VoIP providers these days to handle various NAT based configurations in their border controllers or SIP servers - the first troubleshooting step from their side would often be to *disable* any nat helpers. If that's no good, try restricting each ATA/phone to a different RTP port range and port-forward them. None of the STUN etc that have arrived, since I last looked, any good? Thanx. *** NOTE *** Please DO NOT CC me. I am subscribed to the list. Mail to the sender address that does not originate at the list server is tarpitted. The reply-to: address is provided for those who feel compelled to reply off list. Thankyou. Rod/ --- This life is not the real thing. It is not even in Beta. If it was, then OpenBSD would already have a man page for it.
