If the underlying IP network is unreliable, then packetized voice calls will be unreliable. For some, the internet is an acceptable transport for voice. But there’s no guarantee of reliability across internet networks.

There are many customers that enjoy the benefits of voice-data convergence on their LAN, but use dedicated circuits (T1 or POTS) to connect their business to the outside world. It’s easy to guarantee VoIP performance when you own the LAN infrastructure — no so on the hit-or-miss internet.

Rod Montgomery
Director of Services, Digium, Inc.

Good post! I’ve done this myself many times.

I too thought that TFTP had to be on the same subnet, and was not routeable. However, a couple of people who ought to know have corrected me on this recently. Are we sure that they absolutely must be on the same subnet?

Of course, when provisioning phones remotely you’d use FTP or HTTP over TFTP.

There are two quick security-related conclusions to draw from this demonstration: long term, service providers should adopt a security model that does not rely on the good behavior of their customers; short term, service providers that get burned by such abuses might respond by treating PBX endpoints as untrusted, which will limit their utility.

Still, this is a good example of the unique power that Asterisk brings to telephony solutions — and a big reason so many new products have Asterisk under the hood.

BTW, thanks for the tour on Canada Day. I enjoyed visiting with you and Garrett.

All the best,
Rod Montgomery
Director of Services, Digium, Inc.

I have never experienced any echo problems with the spa-3102 in that type of scenerio or for that matter cross talk. The only real difference between the spa-3000 and spa-3102 is that the 3102 can act as a router where the spa-3000 is a LAN device. The firmware on these units are more up-to-date than the spa-3000 so that may have something to do it. Hope that helps.

I am getting instructions created as we speak and will update everyone on how to obtain them when available. I will like to note that when these instructions become available, they were tested and proved working with a PSTN landline (NOT using an analog PBX connected to station line) as this is another common application for this device. Like I said I will update everyone when this becomes available.

Thanks,

Chris Heinrich

The two devices talk to each other via IP addresses. It is strongly suggested when using these devices in a point to point (spa-3102 to spa-3102 pure VOIP) and back to back (spa-3102 over VOIP to spa-3102 then out PSTN) that you set each unit’s IP address statically. Using them within a VPN or LAN environment cuts down on NAT issues and one way audio issues, and even placing both units on “public” static IP addresses provided by your ISP is suggested. There is often a charge associated with this however. And you can also use this device as an external FXO gateway to your Asterisk server as well.

Can you explain how to set SIP ports on modern popular hardware phones such as the medium priced (or are they entry level these days) Sipura/Linksys/Cisco line?
Why would you not use 5060? If you have several phones behind NAT on the same LAN, is there a logical way to set these? How does the other endpoint see this? Enquiring minds and all that…

I shall wait here on ICE for a STUNning discussion in a future article.