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? No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52
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?
No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52
_______________________________________________ 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
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?
No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52 _______________________________________________ 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
_______________________________________________ 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/ 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
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? No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52 _______________________________________________ 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 _______________________________________________ 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/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1515 - Release Date: 23/06/2008 19:16
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?
No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52
_______________________________________________ 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
_______________________________________________ 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/
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
No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1515 - Release Date: 23/06/2008 19:16
_______________________________________________ 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/ 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
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?
No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52
_______________________________________________ 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
_______________________________________________ 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/
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
No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1515 - Release Date: 23/06/2008 19:16
_______________________________________________ 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/
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
-- 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
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? No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52 _______________________________________________ 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 _______________________________________________ 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/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1515 - Release Date: 23/06/2008 19:16 _______________________________________________ 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/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1516 - Release Date: 24/06/2008 07:53
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?
No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52
_______________________________________________ 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
_______________________________________________ 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/
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
No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1515 - Release Date: 23/06/2008 19:16
_______________________________________________ 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/
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
-- 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
No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1516 - Release Date: 24/06/2008 07:53
_______________________________________________ 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/ 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
Correct. The conference is unusable. When 2 people in the conference they can not hear each other or can hear each other intermittently. I’ve played with the energy level and flags settings but without any luck. The pcap I’ve done is on the client. When doing pcap on the FS box I get the same exact result (corresponding to the client side) however I get completely different result from my other client (an IP phone on which I can not run pcap). The IP phone doesn’t seem to send any RTP when in conference and does send when not in conference. I’m still trying to figure out the difference between the two FS builds (FC4 and v1) but I have a question: Did you introduce the CN (comfort noise) feature after RC4 or was it there before? I’m quite lost with this now and it leaves us with RC4 the only viable production candidate for now. Any assistance will be appreciated. We’re now building another FS box on a different machine and we’ll try to reproduce it there. _____ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, June 25, 2008 8:31 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Recent VAD enhancement? 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? No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52 _______________________________________________ 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 _______________________________________________ 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/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1515 - Release Date: 23/06/2008 19:16 _______________________________________________ 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/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1516 - Release Date: 24/06/2008 07:53 _______________________________________________ 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/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1516 - Release Date: 24/06/2008 07:53
To add to this we have noticed the same with some Dlink Phones (DPH-120s). The audio 'cuts out' intermittently with CNG enabled on the server side, very noticeable in conferences (though they are still usable). The phone will refuse to send any RTP (no network activity for 0.5seconds). This does not happen with Grandstream, Quintum, Cisco, Polycom, Linksys phones. The phone does not do this with asterisk since there is no CNG support in asterisk. I dont think your problem is related but just trying to add to the wisdom :) UV wrote:
Correct. The conference is unusable. When 2 people in the conference they can not hear each other or can hear each other intermittently.
I’ve played with the energy level and flags settings but without any luck.
The pcap I’ve done is on the client. When doing pcap on the FS box I get the same exact result (corresponding to the client side) however I get completely different result from my other client (an IP phone on which I can not run pcap). The IP phone doesn’t seem to send any RTP when in conference and does send when not in conference.
I’m still trying to figure out the difference between the two FS builds (FC4 and v1) but I have a question:
Did you introduce the CN (comfort noise) feature after RC4 or was it there before?
I’m quite lost with this now and it leaves us with RC4 the only viable production candidate for now. Any assistance will be appreciated.
We’re now building another FS box on a different machine and we’ll try to reproduce it there.
------------------------------------------------------------------------
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Wednesday, June 25, 2008 8:31 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Recent VAD enhancement?
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 <mailto: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> [mailto: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 <mailto: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 <mailto:anthony.minessale@gmail.com>> wrote:
what happens when you call 9999?
2008/6/23 UV <uv@talknet.com.au <mailto: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> [mailto: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 <mailto: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 <mailto: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?
No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto: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
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto: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/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1515 - Release Date: 23/06/2008 19:16
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto: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/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
No virus found in this incoming message. Checked by AVG.
Version: 8.0.100 / Virus Database: 270.4.1/1516 - Release Date: 24/06/2008 07:53
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto: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/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1516 - Release Date: 24/06/2008 07:53
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-- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz
I have tried this on my windows VM and it was fine. You're saying you don't even get music when someone is sitting idle? I just tried it again just now before I sent this email. /b On Jun 24, 2008, at 11:52 PM, UV wrote:
Correct. The conference is unusable. When 2 people in the conference they can not hear each other or can hear each other intermittently.
I’ve played with the energy level and flags settings but without any luck.
The pcap I’ve done is on the client. When doing pcap on the FS box I get the same exact result (corresponding to the client side) however I get completely different result from my other client (an IP phone on which I can not run pcap). The IP phone doesn’t seem to send any RTP when in conference and does send when not in conference.
I’m still trying to figure out the difference between the two FS builds (FC4 and v1) but I have a question: Did you introduce the CN (comfort noise) feature after RC4 or was it there before?
I’m quite lost with this now and it leaves us with RC4 the only viable production candidate for now. Any assistance will be appreciated. We’re now building another FS box on a different machine and we’ll try to reproduce it there.
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org ] On Behalf Of Anthony Minessale Sent: Wednesday, June 25, 2008 8:31 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Recent VAD enhancement?
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?
No virus found in this outgoing message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1513 - Release Date: 22/06/2008 07:52
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
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No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1515 - Release Date: 23/06/2008 19:16
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1516 - Release Date: 24/06/2008 07:53
_______________________________________________ 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/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 No virus found in this incoming message. Checked by AVG. Version: 8.0.100 / Virus Database: 270.4.1/1516 - Release Date: 24/06/2008 07:53
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Brian West sip:brian@freeswitch.org
participants (4)
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Anthony Minessale
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Brian West
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Faraz R. Khan
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UV