Peer to Peer SIP Telephony – Will it Work?

That is the question…

I’ve struggled with voice over IP Telephony because I’m so skilled in the old way, that if I brainwashed my mind with the “new way” then I would be selling myself to the devil.

I’ve discussed my frustrations with VOIP from Cisco CallManager Express, to the lousy and vague Asterisk, to the firey but easier to use IP Office from Avaya. In 2011 to 2012, I thought there had to be something easier. Something similar to the ComKey without needing much hardware.

P2P SIP Background

P2P SIP exists, there is a movement as of late, but the idea goes back a decade ago with now vintage and end of life products. Once upon a time there was the  Aastra Venture IP (an IP version of a similar named product), and Avaya’s One-X Quick Edition. These phones were mini servers, that would connect over the network via DHCP, and they would talk to one another and when they would discover each other on the network using a dummy domain that only would be used on an intranet. Configuring the phones would required using the telephone’s menu function on the telset, or using a web browser by using the phone’s IP address.

Connecting to the outside world would be done using a PSTN gateway. Avaya and Aastra sold two gateways for analog POTS and ISDN/T1 trunks. Avaya’s sets from research could be programmed into third party gateways like Cisco routers and alike if the customer had a Cisco router.

Relevant History 

These are called Key Service Unit Less phones. Key Systems were small end phone systems and by the late 1980s into the early 1990s, the advancement of electronics enabled POTS telephones to talk to other phones on the same circuit that couldn’t be done before. Using low level circuitry could do almost the same thing as a traditional phone system. There is some engineering for this to work properly, especially if you’re maxing out to the four lines and sixteen stations.

The Avaya One-X QE and Venture IP sets can only work as these specific P2p phones, so they cannot work on any other SIP switching system other their own inside the phone. If you think about it SIP is like the PC like old office telsets are to dummy terminals. It’s much better to have a phone that can support any SIP protocol outside the phones so if you do acquire some softswitch, it’s not a writeoff. (Avaya’s does have the possibility to flash out the ROM but because it’s EOL for their early VOIP sets, that will not ever happen.)

Use case:

There was a need for a SIP P2P set for a VIP-type of authority. The environment was switching to IP Telephony, and SIP is a mandatory requirement in case the ITSP or SIP PBX goes down. Especially during emergencies.

I did this in a home lab, because in 2016 I decided to use my Definity PBX as the primary telephone system; this enabled me to use some SIP sets I acquired from a buyback deal in 2016 and some Polycom sets I found off a business’s front lawn during the same time too. A couple Mitel 5220, 5224, 5330 IP sets were also tested.

Debunking myths

The theory was:

  1. How could the phones find each other when making a call?
  2. What types of settings do I tell the phones to go to (Proxy addresses, etc)?
  3. Do I need to have a proxy address?
  4. What type of benefits do I get having a basic, and a multi line telephone system?

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