Hi,
I have been testing various Polycom phones with the wideband codec G722.
Here is the result of my research.
--
Goals
1. two phones can call each other with HD when available
2. an HD phone will fallback to G711 when G722 not available
Platform:
Thirdlane MTE with Asterisk 1.4 and PBX Manager 6.1.1.6.
Try #1: canreinvite
canreinvite: yes
nat: no
g722 added to codecs list on test extensions.
canreinvite:
Asterisk will not issue a reinvite if NAT is detected, OR if the 't' option is used in the Dial command.
There is undoubtedly a Thirdlane setting that has 't' turned on.
This is not an option because then we cannot transfer calls - or would this only impact star-code transfer and not transfer from the Polycom buttons?
http://fonality.com/trixbox/forums/trixbox-forums/open-discussion/canre…
http://www.voip-info.org/wiki/view/Asterisk+sip+canreinvite
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Try #2: force codec to g722 only
This works from 1 phone to another so long as g722 is the only codec allowed in asterisk on both phones.
However, attempt to call from a phone not so configured, results in "the person at extension xxx" is on the phone,
as Asterisk must be rejecting the call.
Try #3: force codec, but have g722 first and g711 2nd
have g722 be the first codec in the list
This works better, phones can call each other SO.
But If g722 is not in a phone's allowed codecs list, then a call from HD to non-HD fails
g722-enabled phone that has g722 first in the list, cannot call out to the PSTN.
It looks like Asterisk will not renegotiate or redo codec selection once it's been chosen on each call leg. There's a patch floating around for Asterisk 1.4 to make this smarter, but who wants to do that.
So it seems that the only option may be to use Asterisk 1.6, which has a transcoding codec for G722 built-in.
Has anyone successfully used Thirdlane MTE with Asterisk 1.6 and polycom phones with the G722 wideband codec?
I searched the forums for G722 but found only one reference with no information in it.
Thanks in advance,
Jawaid
Looks like libresample is on
Looks like libresample is on the MTE 2.0 version by default. Or did you mean I need a different one?
no, theres only one
no, theres only one libresample. Its needed to build the codec_resample.so module
you also need libresample because the sample rate between wideband and narrow band are at different sample rates.
asterisk does not do codec negotiate the way you think it will.. if phone A is set in order of preference G722,ulaw and phone B is just ulaw, phone A is still going to communicate with asterisk as G722 and you will transcode/resample between the two phones.