you can set the global var send_silence_when_idle=true in vars.xml On Mon, Jul 27, 2009 at 4:23 PM, Keith Laaks <keithl@voxtelecom.co.za>wrote:
Hi All,
I am testing a range of G722 capable DECT based CPE.
With one range, I have noticed that the first 200ms or so of each separate prompt file being played back is played out distorted from the DECT handset.
When having a normal conversation, the quality is excellent, but when accessing your vmail, all the individual audio files making up the menu choices exhibit the distortion, which is pretty annoying.
The same unit using G729, alaw or ulaw works 100%.
I wonder if anybody else has uncounted this issue?
My guess at this point –
There may be a short break in the RTP between the separate files being played out by FS that makes up any menu.
During this time the DECT handset’s AGC probably goes to MAX amplification (as its not receiving any input during the short break in RTP).
Then, when the RTP returns at the start of the next file, the AGC boosts the audio into clipping zone and takes 200ms to dampen down back to normal good levels.
Looks like in these devices the G722 encode/decode is actually done in the DECT handset and not the voip-base unit.
Is there any parameter that can be set in FS to ensure that the RTP keeps flowing, sending ‘silence’ between prompts ? Would be interesting to validate the above ‘guess’.
Best Regards
Keith
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400