Is there a FAQ file? 'Cause I have lots of Qs

Hi,

I have the most basic questions. I also have a Sipura SP841, an account on Broadvoice, and no phone service. ;) I'd like to trade in the questions for some answers/knowledge and maybe even, as a bonus, a working VOIP system.

Basically, the 841 is here behind my Linux NAT/firewall (netfilter) host and my net connection goes through Comcast cable. The phone has a fixed

192.168 address and the firewall uses DHCP with Comcast. I can see the phone's menu from browsers on other machines on my network, but I can't call out and incoming calls go straight to voice mail with the "Joe is busy" message.

Netfilter is set to ACCEPT udp from the external interface on ports 69,

5060:5063, and 10000:20000.

Is there a FAQ for this newsgroup? A good source for the very rudiments of application? I don't mind starting from the start to learn this stuff. Obviously jumping right in wasn't the optimal approach.

TIA,

Joe

Welcome to the "SIP+NAT Nightmares Club" :-)

OK, here's the deal. The SIP protocol is very NAT-unfriendly, for basically two reasons:

1. As it name says, it only deals with session initiation; the actual voice packets are transported with a separate protocol called RTP, which uses UDP but with different port numbers. Even when the SIP issues are ironed out, sometimes one can't hear audio due to RTP problems.
2. SIP messages contain, ASCII-encoded, several parameters that are affected by NAT: IP addresses and port numbers (both for SIP and RTP).

Some relief comes from enhancements made to the protocol and additional helper protocols, such as STUN

allows a NATted client to find out the external address of its router (so that it may communicate to the peer THAT address, and not its own on the LAN) and the type of NAT (to attempt educated guesses on port mappings). Still, it's a struggle. To some extent, this is a repeat of the headaches with FTP behind NAT, the attempts at fixing them with "Passive FTP" etc. Only, worse.

I'm not familiar with the SPA-841 but I have used SPA-2100 and SPA-3000, and I think most settings are similar. In my SIP settings, under "NAT Support Parameters" I have set all parameters to "Yes" except "Send Resp To Src Port:" which is "No". As STUN server I use stun.fwdnet.net:3478 . "NAT Keep Alive Intvl:" is set to 15.

Then, in the "Line 1" and "PSTN" screens, under "NAT Settings" I have set to "Yes" both "NAT Mapping Enable:" and "NAT Keep Alive Enable:"; the "NAT Keep Alive Msg:" is better left empty rather than the default "$NOTIFY". And I recommend you also to set "Symmetric RTP:" to "Yes", otherwise you might lose audio in one direction, or both.

Also, for anything related to Sipura check out the excellent site

and its forums, which are moderated by competent people.

Good luck,

Enzo

Enzo,

Thanks for taking time to help me.

Do you know if there are specific settings I need to use for the NAT machine? The only change I introduced for the phone was the ACCEPT udp on the 69, 5060:5063 and 10000:20000 ports.

I've made these changes as I see them on the SPA-841 screens. I've progressed from "Not Registered" and no lights on the unit to "Registration Failed" and all the lights on the unit glowing amber. Considering where I started, I call that progress!

I also found some settings from the config tool, but there is some trouble because when I try to upload those settings I get an error on the config page in the browser.

I'd be better at debugging if I had some idea how this is supposed to work. When I debug a web app, I understand basics like http, rmi, etc. I'm poking at this like a voodoo doll, trying to get the right combination of pins to effect my desired result. Not exactly an optimal process, eh? ;)

Thanks,

Joe

This should be OK depending on the SPA-841's settings. Usually each subunit (e.g., each FXS port in a SPA-2100) has its own SIP port, and the SIP ports start from the default 5060, so you should be in good shape for that. For the RTP ports, the range is specified somewhere in the Sipura setup web screens. For example, in the SPA-3000 there are two fields in the "SIP" screen labelled "RTP Port Min:" and "RTP Port Max:" (in my case, I set them to 16384 and 16482 respectively, which falls inside the range you gave to tour router). But anyway RTP gets in the picture at a later stage, after the session initiation, so this is not what makes your registration fail.

Are you sure your username and password are correct? Why don't you test the account first with a softphone, such as Xten X-Lite or the simpler Firefly

, under "Download the latest third-party version of the Virbiage Softphone here."

Which error, exactly? And have you used a tool specific to the SPA-841? As far as I know, tools that work for SPA-2000 / SPA-2100 probably won't work for the SPA-841.

If you want a quick tutorial about SIP, you may try this one, which helped me a lot when I began to deal with SIP):

It should at least give you a rough idea of the "big picture". Then you might get Ethereal and use it to sniff the UDP packets to understand why the "REGISTER" transactions fail. The biggest problem is that the PC running Ethereal (placing the LAN card in "promiscuous mode") should be able to "see" the packets between SPA-841 and SIP proxy, but in these days LANs are almost all switched... You should find an old non-switching hub, or configure the PC as a router and put it in the path between the SPA-841 and the Internet, etc.

Cheers --

Enzo