Joe Abley wrote: There's no useful way to use H.323 through a NAT though, at least that I have seen working.
In enterprises this has never been a problem as H.323 works fine over any kind of tunnel that goes over NAT and that's already there for other purposes (VPN for example). I have multiple h.323 VOIP phones at home over one IP address, one that is actually IPSEC encrypted. As of SIP, Vonage does indeed use it so are many other SIP shops and it works fine over NAT Vonage or not Vonage. I am not that familiar with the Cisco ATA-186 (too pricey) but on other SIP phones such as the popular $70 Grandstream if you know the IP address and port of the remote SIP phone you want to join you can dial it directly. Michel.
I predict the next generation of VOIP to: a) run over encrypted tunnels. b) have a server based out of the US.. c) meet massive resistance from the EffBeeEye.. a) will change the issues being debated here. -- A host is a host from coast to coast.................wb8foz@nrk.com & no one will talk to a host that's close........[v].(301) 56-LINUX Unless the host (that isn't close).........................pob 1433 is busy, hung or dead....................................20915-1433
On Sat, 2003-11-01 at 11:57, Michel Py wrote:
but on other SIP phones such as the popular $70 Grandstream if you know the IP address and port of the remote SIP phone you want to join you can dial it directly.
Michel.
We use the Grandstream via sipphone.com for office to office calls. It is using the RTSP. Just doing some cheap testing before we integrate this into our Soft Switch, PBX and the PSTN. The Sipphone has a "STUN" server function that makes doing SIP behind NAT/PAT workable. I am a little hazy on its function as I am testing and wanted the phones on public IP's. Seems to keep the NAT/PAT translations constant by communicating with a remote server. Some users are doing SIP with this phone without problems behind NAT/PAT. -- James Edwards Routing and Security jamesh@cybermesa.com At the Santa Fe Office: Internet at Cyber Mesa 505-988-9200 SIP:747-669-1965
james writes on 11/1/2003 2:30 PM:
The Sipphone has a "STUN" server function that makes doing SIP behind NAT/PAT workable. I am a little hazy on its function as I am testing and
perhaps short for secure tunnel - an ssl tunnel that takes your sip traffic through http or something, and proxying them through a remote server? srs -- srs (postmaster|suresh)@outblaze.com // gpg : EDEDEFB9 manager, outblaze.com security and antispam operations
On Sat, Nov 01, 2003 at 03:15:18PM -0500, Suresh Ramasubramanian wrote:
perhaps short for secure tunnel - an ssl tunnel that takes your sip traffic through http or something, and proxying them through a remote server?
Simple Traversal of UDP through NAT, for details see: http://www.ietf.org/rfc/rfc3489.txt and more generally http://www.ietf.org/html.charters/midcom-charter.html Bill. - - - Bill Owens Manager, Network Development NYSERNet, Inc.
Has anyone been experiencing connectivity issues with Wiltel over the last 12 hours? -brian
On Sat, 2003-11-01 at 14:30, james wrote:
We use the Grandstream via sipphone.com for office to office calls. It is using the RTSP. Just doing some cheap testing before we integrate this into our Soft Switch, PBX and the PSTN.
The Sipphone has a "STUN" server function that makes doing SIP behind NAT/PAT workable. I am a little hazy on its function as I am testing and wanted the phones on public IP's. Seems to keep the NAT/PAT translations constant by communicating with a remote server. Some users are doing SIP with this phone without problems behind NAT/PAT.
On mine I'm using for testing, if you turn NAT traversal on, and erase the contents of the STUN field (don't leave them at 0.0.0.0, it'll break), it does NAT traversal if the server supports it, without need for a STUN server, which I still can't find a copy of. Fortunately it's unnecessary. It works, as long as I don't try to contact another phone behind another NAT. -Paul -- Paul Timmins <paul@timmins.net>
for a STUN server, which I still can't find a copy of. Fortunately it's unnecessary. It works, as long as I don't try to contact another phone behind another NAT.
That is the very essence of why I think NAT in the long run is a bad idea... What good is a phone that can't contact another phone. One of the main advantages to VOIP is that you can achieve some level of provider independence and still have phone service. Sure, you need some level of ISP, but, you can have more than one of those and SIP still works when one is down. If you are dependent on a particular company running a proxy, then, if they hose their stuff, you're out of luck. Owen -- If it wasn't signed, it probably didn't come from me.
participants (8)
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Bill Owens -
Brian Boles -
David Lesher -
james -
Michel Py -
Owen DeLong -
Paul Timmins -
Suresh Ramasubramanian