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:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
