so the conference is never working you are saying? When 2 people are in the conference can they hear each other? are you doing pcap on the FS box or on the client side? There is energy detection in the conference that you could try turning down in the conference profile. <param name="energy-level" value="0"/> try this setting to make the conference send audio at all times <param name="member-flags" value="waste"/> On Tue, Jun 24, 2008 at 1:38 PM, UV <uv@talknet.com.au> wrote:
9999 is pretty much the same. I can't tell if 90% or 70% of the traffic is lost but something definitely makes the FS stops sending traffic and start again (as if it was VAD).
The client sends RTP as usual. No change there. The loss is only from the FS server side.
Problem persists when adding callers to the conference. Callers sending RTP to the server but not receiving much back.
I've double checked and there's no problem when dialing from one extension to another. RTP stream is fluent.
It's on one box we tried. Only tested on windows. That's why I'm asking if anyone had that too.
Again, this phenomenon is predominately noticeable for MoH. It doesn't seem to occur when playing a file, generating a tone or streaming mp3.
There's a different "disturbance" when playing WAV files but I'm not sure it's related (also doesn't happen on the RC4). The audio seem to skip forward from time to time – like short bursts of fast-forward effect occasionally.
Playing MP3 sounds good.
------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Wednesday, June 25, 2008 2:39 AM
*To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Recent VAD enhancement?
so are you also saying that the client is not sending RTP either? did you try adding subsequent callers to the conference? is this on more than one box? is this only on windows?
On Tue, Jun 24, 2008 at 11:37 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
what happens when you call 9999?
2008/6/23 UV <uv@talknet.com.au>:
I've checked the packet flow in wireshark and it seems like the FS is sending RTP for about 2 seconds (the "alone announcement") and then stops sending RTP. From time to time few packets arrive and you can hear the MoH momentarily.
The call scenario was a softphone (SJPhone) calling 3011 conference room. Both are on the same LAN speaking directly.
There were no error messages.
Same scenario on RC4 works fine.
No Asterisk involved, installed or present in the surrounding area.
Any ideas?
------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Tuesday, June 24, 2008 7:14 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Recent VAD enhancement?
check the packet flow in wireshark also
On Mon, Jun 23, 2008 at 2:06 PM, Brian West <brian@freeswitch.org> wrote:
By any chance is Asterisk involved in this at any point along the call path?
/b
On Jun 23, 2008, at 1:55 PM, UV wrote:
I'm trying to send this and it doesn't seem to post… here's another attempt:
After upgrading from latest stable version for windows (RC4) to latest version, I've noticed a significant loss of audio during a call.
It almost seems like there's an enhanced VAD that triggers the audio on and off – although the VAD feature is off. Why VAD? Because as you inject audio on the receiving side, you can then hear incoming audio from the transmitting side.
Although this is not noticeable when using tone generation or shout, it's mostly evident when playing MoH (Music on hold, not Medal of Honor). It sounds like about 90% of the audio is gone.
We used identical configuration files for both tests.
We're still investigating this behavior and soon we'll dive to network level and debug outputs.
Anyone else encountered that as well?
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
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
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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