VoIP Users Conference Friday, Nov 7: Wideband Telephony

What’s Happening This Friday?

The VoIP Users Conference weekly call on Friday, November 7, 2008 is all about wideband telephony. That is, using VoIP to deliver call quality vastly superior to the normal public telephone network (PSTN.) Our guest will be David Frankel, CEO of ZipDX, a commercial conferencing service that is specializing in wideband conference calling for businesses.

As usual anyone may join the call which gets underway at 12 noon EST on the Talkshoe conference service. That conference service can be reached by dial-in over the PSTN or by SIP URI. Details on how to connect to Talkshoe by various means can be found here.

For this one call only we will also be using the ZipDX wideband conference bridge. ZipDX and Talkshoe will be connected so that everyone will be on the call. Anyone connected to the ZipDX bridge using a suitable phone will be able to experience the call in G.722-based wideband quality.

Everyone else will experience the normal G.711 based narrowband conference quality that we all know and (despise) love.

Which Phones Are G.722 Capable?

To connect to the G.722 based conference bridge your phone must be capable of using the G.722 codec. Many business class SIP phones support this codec including the following:

  • Polycom: SoundPoint IP550, IP650, IP6000, IP7000
  • Snom: All 3×0 models with recent firmware
  • Cisco: 79xx series
  • Siemens: S675IP, S685IP
  • Grandstream: requires wired headset for improved audio
  • Mitel
  • Avaya
  • Counterpath: Eyebeam Soft Phone (Windows OEM version only, not available for sale online)

What If I Don’t Have One Of These?

If you don’t have a suitable phone then you may still experience the call in wideband. ZipDX has very graciously provided us access to a small quantity of licenses to the G.722 capable version of Counterpath’s Eyebeam soft phone for Windows. These are available on a first-come first-served basis by sending a request to me at mgraves-at-mstvp.com.

How Do I Test To See If My Phone Is Suitable?

If you want to test your endpoint’s connection to ZipDX just dial the SIP URI wbdemo@conf.zipdx.com. This is an automated service that plays various recordings and announces at the beginning if you are connected with a wideband or narrowband codec.

Note that dialing by SIP URI often requires that you create an entry in the phones contact list as many phones only support numeric dialing ala PSTN.

How Do I Reach The Wideband Conference Server?

Now that you know your phone is suitable you can reach the call using ZipDX by the following means:

1) If you are explicitly invited to the ZipDX conference then call the SIP URI 1234567@login.zipdx.com, where 1234567 is replaced with your unique ZipDX PIN. To do this we have to send you an email invitation with a PIN in advance. Your email address must already been added to the conference by the conference admin (that’d be me!)

This approach is best as it lets the conference administrator know who is talking on the web based control panel. It makes the call easier to moderate.

2) If you want to join the call “anonymously” then they can dial the SIP URI 541042@login.zipdx.com. This is a guest login so the conference administrator will not be able to identify you at a glance. As a result you are more likely to be muted when we are trying to silence background noise on the call.

Can Anyone Assist Me?

I will try to be available as much as possible this week to help people get setup for the call. I do this because I truly believe that VoIP quality should surpass that of the PSTN. Further, that everyone who can should have this experience asap. I can be reached at:

  • email: mgraves-at-mstvp.com
  • PSTN: +1 713 481 0387
  • SIP: mgraves@mstvp.onsip.com
  • Skype: mjgraves

The best way to test your wideband capability is using the demo SIP URI at ZipDX as given above. I have several phones logged into my OnSIP account so you may be connected narrowband or wideband depending upon which phone I answer.

Please be mindful that I am on US Central time. If you don’t reach me please leave a message and I will return your call.