2004 April

April 2004


Presence was the big word everyone used at Voice on the Net 2004. But the solutions that was showed can easily be divided in two categories: phone status and user presence.
A lot of corporate PBX solutions display phone status some way. A button says \”Olle\” and the light on the side shows whether I\’m on the phone or not. This is a feature a lot of Asterisk user\’s ask for, and a lot of GUI software add-ons solve this problem.
Instant Messaging Presence on the other hand does not involve my equipment, it\’s really about me, myself and possibly also I. Am I available for chatting? If not, why? There\’s a lot of standardization going on here. Most notably Jabber and SIMPLE, the SIP based Instant messaging platform. Both of these are heading for XMPP, the IETF platform.

The really interesting part is whether the presence you want is my phone\’s status or my status. And my status depends on who\’s asking. My kids, my wife, a vendor, a telemarketer, my ex-girlfriend, my friends – they all will get different answers. My status depends on who you are and a combination of a lot of variables collected from my phone and possibly other devices.
Brian Rosen, one of the IETF SIP architects, pointed this out on his closing keynote at Von. Brian, I\’m looking for a solution, some open soruce code – with or without your megabit video streams…

We need to integrate a presence solution with Asterisk. Asterisk can deliver your phone\’s status, but have no clue on your status. And while thinking about it, who has :-)

The first source code of Pingtel\’s SIP software was released yesterday. Pingtel have uploaded tar archives – \”tarballs\” – of the software to the SIPfoundry archive. This first release is just for developers who want to peek into the code. It can\’t be built. Pingtel is working on removing dependencies to third party software that can not be released as open source.
On the web site, there\’s finally a lot of source code, but it\’s not easy to figure out how it all works together and how to build something from it. Pingtel in the announcement said that their working on a future release that will be a working system without the third party software.

There\’s a lot of buzz of Voice over WiFi. Some people like to compare it with Cell Phones (gsm, 3g), but I don\’t feel we\’re close. Cell phone batteries last much longer than WiFi phone batteries does. WiFi consumes power.
On campuses, I believe WiFi phone will play a larger role. As soon as we have decent WiFi phones, DECT will start to fade away and end up in the same grave as Hiperlan and HomeRF.
Vonage, a broadband telephony provider in the US and Canada, has their eyes set on Voice over WiFi, or VOW. Ed Sutherland reports in Wi-Fi-planet:

While the final handset may change between now and the end of 2004, both Schultz and Tribolet agree on what any Wi-Fi phone will provide: one hour of talk time with four hours of standby. A major failing of past and many current Wi-Fi phones is their heavy power demands. Unlike a cell phone, which can \’hibernate\’ until the next incoming call, Wi-Fi based phones, dependant on the unpredictable nature of Internet packets, must remain ever vigilant — and always on. This need for constant power means the talk and standby times for Wi-Fi phones have lagged far behind alternative wireless handsets.

When registering for Hotsip\’s newsletter, there\’s no way to register a SIP url instead of my telephone number. I guess this is a mistake, but it\’s certainly embarrassing for a company that is supposed to be one of the leaders in the SIP proxy and software client market :-)

Cullen Jennings, active developer in the vovida.org open source project and a Cisco engineer, talked about SIP security and NAT traversal as part of the VON Conference recently in Santa Clara. He has been shopping around for broadband routers with NAT support and tested all of them with STUN to see if and how they supported SIP telephony. The result is published in an IETF draft – very interesting reading. Most NAT devices worked with one IP phone. Some with two and only one didn\’t work at all.

A major issue in working with NAT traversal solutions for various protocols is that NATs behave in many different ways. RFC 3489 (STUN) classifies these and provides a method to test them. This draft describes the results of testing several residential style NATs. Several NATs attempt to use the same external port number as the internal host used. This is referred to as port preservation. On the NATs that did this, some were found to have different characteristics depending on whether the port was already in use or not. This was tested by running the STUN tests from a particular port on one internal IP address and then running them again from the same port on a different internal IP address. The results from the first interface, where the port was preserved are referred to as the primary type while the results from the second interface, which did not manage to get the same external port because it was already in use, is referred to as the secondary type. On most NATs the secondary type is the same as the primary but on some it is different; these are referred to as nondeterministic NATs, since a client with a single internal IP address can not figure out what the type of the NAT is.

\"PicturePingtel in cooperation with Vovida.org and the Reciprocate project launched SIPfoundry at Von 2004. The new organization will host Pingtels PBX software, recently announced to be released as Open Source, vovida.org software and the Reciprocate SIP stack. The new organization is welcoming other open source voip projects to work with SIP foundry. Time will tell how this succeeds.
Vovida.org now have no dependency upon Cisco and will continue as a free project. The Pingtel software is not yet judged by outside developers – will it stand the test and be taken care of by an Open Source community? The competition from Asterisk and SIP Express Router – two of the most successful Open Source VoIP projects – may be a road block. One thing is for sure, both projects were represented at the initial meeting in Santa Clara and they will be going through the source as soon as it is released.

SIPfoundry is about improving and adopting open source projects related to SIP – the Session Initiation Protocol. SIP is about finding ways to create connections. What more could you ask for?

We are just getting started, and expect rapid expansion in both the projects available to users and the services available to developers from SIPfoundry.

I understand that this is a big step for Pingtel and wish them all luck with this new phase of the company. Working out how to survive being a commercial company working with Open Source developmenet is not an easy task. I look forward to downloading the PBX!