On 01/05/2011 01:29 AM, Bruce B wrote:
Hi Everyone,

You should really spend some time learning how to use a widely available search engine, such as Google or Bing. Many of the questions you ask here could be quickly answered that way.

1- Are the Siren7 and Siren14 the G.722 HD voice codecs?

That depends on what you mean by "the G.722 HD voice codecs". G.722 (no suffix or annex) is a specific voice codec, and it is unrelated to Siren7 and Siren14. G.722.1 is the same thing as Siren7, and G.722.1 Annex C is the same thing as Siren14. These are fairly well explained on the Wikipedia pages for G.722 and G.722.1.

2- Are these codecs only for Polycom units or are they universal across
all other SIP phones that advertise the HD voice codec like Aastra?

There is no "the HD voice codec". The most widely available HD voice codec is G.722, but there are others available as well in endpoints from various manufacturers. G.722.1 and G.722.1C are available in many Polycom SIP devices, but not yet widely available in other devices (although there are softphones that have them), although they are available as binary add-on modules for Asterisk.

3- What is the main difference between the two and is it advisable to
run these over the INTERnet (not INTRAnet)?

Which "two" are you referring to here?

G.722 is frequently used over the Internet (the weekly VUC makes it available, for example), and G.722.1/C use less network bandwidth than it does, so they should be usable in the same situations as well. Whether they are advisable for you to use or not depends entirely on your network connectivity, bandwidth and packet loss situation. None of them use more network bandwidth than G.711, though, so if your network connections can already handle G.711 calls without causing unacceptable audio disturbance, you should be fine using any of these codecs as well.

Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

Reply via email to