Sjur Usken

Views on new technologies and business opportunities from Sjur Usken

Monthly Archives: June 2008

Review of Polycom 6000 and 7000 conference phones

Just had two phones landing on my desk, the new conference phones from Polycom. The Polycom 7000 looks great and stylish, but the 6000 is rather dull. Both have the same great codecs, so there should not be a big difference on the sound quality.


The 7000 does have extra input like USB and micro-jack (2,5mm) for mobile phones.
It also seems the 7000 is the only one supporting Power over Ethernet (PoE). There were no power to it at all in the box, only a network cable. Luckily I had lots of single PoEs laying around…

There gotta be a bug in the setup through the web interface. I could not make it set a authorization name/number. I also saw it in the SIP code that it did not send any authorization user in the digest authentication. I tried several times through the web interface, but ended up a day later putting it directly into the phone through the built in screen. Then it worked…. I did read the manual, but this is missing the middle part. It is good on user behavior (volume up and down) and good on setting up an automatic provisioning system (APS). But I was just going to put in one single account.

Finally, the usage

Great microphones! It picked up sound from all over the room. We had music playing, others discussing nearby but could still carry out the conversation! Say goodbye to lousy ISDN quality!! I really wonder how they actually make it work in full duplex, we were chatting in both directions with no problems. I have now wrapped up the 6000 (the ugly one..) and shipping it 2000 kilometers north to our support center. We have regular meetings, so that will be a good test. We used a Linksys 942 with loud speaker last time, and it was dreadful sound. The one we normally use is the Grandstream GXV-3000 with video, this works excellent as long as you have an extra conference “head-set”.

I would absolutely go for the Polycom 7000, but then I haven’t heard the price yet….


OpenSBC and other nice projects

The Session BorderController is a VoIP specific transformer. It is really a mock up to handle the differences between vendors and bad implementations of VoIP protocols. Some SBCs also do transcoding and SIP to H323 conversion, which is functional. But in my opinion a SBC is unnecessary in a perfect world.

You can argue that a firewall has a SBC built in, which is correct. SBC can be called a protocol specific firewall. And that costs money! I have checked prices on Covergence, Juniper, InGate and lots of others. One thing in common, it costs! It costs to be an early VoIP adopter, paying extra to get everything to work together! So actually paying for lousy work done by others implementing the SIP stack….

Then it is nice with OpenSBC, FreeSwitch and similar projects. They provide means to get the different vendors to work together, making you able to choose between different vendors. I wish the open source community all the best! Let’s make this great togehter!

Let the phones do it themselves!

Finally some “out-of-the-box” thinking. Avaya has launced their Quick Edition phones. When the individual devices are getting more and more memory and CPU, why not let them also do the job?

It is as easy as genius. If you need more phones, just add more. No central PBX needed. All the phones are stand-alone and manage themselves through some sort of local P2P. And you get quite a bit of functionality as well. Some of the features (from )

  • Voicemail Backup
  • Greetings and Prompts
  • Message Waiting Indicator
  • Redirect to Specified Extension
  • Telephone User Interface
  • Visual Voicemail
  • Message Monitoring
  • Message Sorting
  • Callback from Message
  • Administration

What will be the next big thing? We’ll just have to create it ourselves!


VoIP for the 3rd world

The VoIP revolution could make businesses more effective in the Western world, but maybe the major revolution will be in the third world countries.

David Rowe has been setting up very low power ATA adapters usable for the 3rd world. The adapter can run on only 3 Watt! When you combine this with a Linksys WRT54GL for the backbone, you suddenly have infrastructure for the 3rd world!

The WRT54GL should be set-up as a full mesh network and then we only need P2P SIP! Does anybody know if the OLPC laptop supports a VoIP client? I believe that information is the key to help people out of poverty.


Finally, wide-band sound will benefit VoIP

Have you ever heard “ohh, VoIP, isn’t that bad sound quality?” Personally I’m quite tired of it and VoIP is now ready for the larger market!

Polycom 7000The good news is all the new products with extremely good sound! Polycom has released their Polycom 6000 and 7000 (BTW, why do the need so many digits in their product name…) Once you have started using these, you really don’t want to go back. These solutions will help accelerate the VoIP convergence.

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Callweaver, freeswitch and others are gaining speed!

The VoIP community is gaining speed, with good projects like Callweaver and Freeswitch! It is good to see that projects can evolve without the proprietary timers needed for the Digium version of Asterisk. It is fully understandable that Digium wants Asterisk to be dependent on their phone cards, but it hinders the VoIP development.

T.38 is important, even though the fax machine should have been replaced by a PKI system where you could send authenticated mails. It would also save the environment for prints…

The Callweaver (previous openpbx) can also be used as an session border controller. The SBC is really just a firewall, but with specific rules and algorithmes for VoIP. It is a product for the time beeing until the VoIP standards are fully complied by the different vendors. The SBC market will exist as long as all the vendors follows their “own” implementations of the standards.

Freeswitch is not based on Asterisk, but tried created better with the faults and drawbacks of Asterisk in mind. It can be used as a simple switching engine, a PBX, a media gateway or a media server to host IVR applications using simple scripts or XML to control the callflow. FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP, TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.

I’m positive to the different flavors and distributions to Asterisk and hope they can learn from each other and bring the world forward!

Good job!