I tested two client soft phones (using Counter Path Eyebeam) via the Asterisk 1.4.1 release.
Should both clients only be set with H264 and G729 codecs, calls fail.
Should the clients be configured to also allow the ulaw or alaw codecs, the calls still fail.
Only by moving the G729 codec from within sip.conf below the other codecs, the calls are successfull.
The clients do have native G729 support as well as other hardware UA’s that we have installed. On Asterisk I did not notice a G729 codec within /usr/lib/asterisk/modules, yet believe that Asterisk does support pas thru of G729…
Is there a G729 codec for Asterisk?.. (to be compiled in allowing native support on Asterisk - /usr/lib/asterisk/modules)
We simply wish to standardize on the best quality - lowest bandwidth usage - audio and video codec’s. We do not wish to allow any other “older and/or less efficient” codec usage. (Any experienced input will be greatly appreciated) 8) All additional features such a voicemail, call conferencing, billing will need to be supported as this will be added in due time.
Also, I noticed the following within the asterisk console which I am currently not exactly sure as to the source of the problem:
NOTICE:rtp.c:1254 ast_rtp_read: Unknown RTP codec 124 received from 'xxx.xxx.xxx.xxx’
NOTICE <- same stuff
NOTICE:rtp.c:1254 ast_rtp_read: Unknown RTP codec 126 received from 'xxx.xxx.xxx.xxx’
NOTOCE <- etc
It seems as if I just can’t find any reference to the same issue within the forum…
Also, how does one resolve the codec nr to a specific codec…?
(Ie codec 126 = GSM & codev 124 = h263…