Hi, I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag. CPU usage on the machine is nil. Any ideas? Thanks, -Clint -- Clint Popetz http://42lines.net Scalable Web Application Development
On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint@42lines.net> wrote:
Hi,
I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag. CPU usage on the machine is nil. Any ideas?
Do you mean that when you speak, it takes 10-12 seconds before the audio is heard by someone else in the conference? Also, can you try the same exercise with a soft phone like x-lite? I'm curious to know if this happens only on Skype calls or on any calls made to a conference. -MC
On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint@42lines.net> wrote:
Hi,
I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag. CPU usage on the machine is nil. Any ideas?
Do you mean that when you speak, it takes 10-12 seconds before the audio is heard by someone else in the conference?
Correct.
Also, can you try the same exercise with a soft phone like x-lite? I'm curious to know if this happens only on Skype calls or on any calls made to a conference.
Sip is a whole other can of worms that's not working either. When connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo fine, but it won't listen to DTMF numbers to change the menu. When running xopier on linux, I can't hear anything and DTMF doesn't get through. Both those softphones work fine with asterisk, including dtmf. I was punting on getting sip working correctly in freeswitch until I determined whether skypopen solved the conferencing woes I'm facing with MeetMe/asterisk. (It's entirely possible/probably I just have sip misconfigured...I'm just using the default configuration in that regard.) -Clint -MC
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-- Clint Popetz http://42lines.net Scalable Web Application Development
Try calling the public conference at sip:888@conference.freeswitch.org<sip%3A888@conference.freeswitch.org> that is a properly setup box and can accept any registrations. The sip and the conference all work fine so it's for sure a problem on your end. On Fri, Apr 2, 2010 at 2:04 PM, Clint Popetz <clint@42lines.net> wrote:
On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc@freeswitch.org>wrote:
On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint@42lines.net> wrote:
Hi,
I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag. CPU usage on the machine is nil. Any ideas?
Do you mean that when you speak, it takes 10-12 seconds before the audio is heard by someone else in the conference?
Correct.
Also, can you try the same exercise with a soft phone like x-lite? I'm curious to know if this happens only on Skype calls or on any calls made to a conference.
Sip is a whole other can of worms that's not working either. When connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo fine, but it won't listen to DTMF numbers to change the menu. When running xopier on linux, I can't hear anything and DTMF doesn't get through. Both those softphones work fine with asterisk, including dtmf. I was punting on getting sip working correctly in freeswitch until I determined whether skypopen solved the conferencing woes I'm facing with MeetMe/asterisk.
(It's entirely possible/probably I just have sip misconfigured...I'm just using the default configuration in that regard.)
-Clint
-MC
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-- Clint Popetz http://42lines.net Scalable Web Application Development
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-- 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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
On Fri, Apr 2, 2010 at 2:17 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
Try calling the public conference at sip:888@conference.freeswitch.org that is a properly setup box and can accept any registrations.
Yup, that works, as does calling my existing asterisk pbx.
The sip and the conference all work fine so it's for sure a problem on your end.
Well, this is with a fresh install using: cd /usr/src ; wget http://www.freeswitch.org/eg/Makefile ; make make all cd freeswitch.trunk make install make cd-sounds-install make cd-moh-install the only thing I'v changed is to add <load module="mod_skypopen"/> and <action application="set" data="skype_add_outband_dtmf_also_when_bridged=true"/> and to install skype and make it listen to skypopen. The rest is the default config. -Clint
On Fri, Apr 2, 2010 at 2:04 PM, Clint Popetz <clint@42lines.net> wrote:
On Fri, Apr 2, 2010 at 1:25 PM, Michael Collins <msc@freeswitch.org> wrote:
On Fri, Apr 2, 2010 at 9:38 AM, Clint Popetz <clint@42lines.net> wrote:
Hi, I'm new to freeswitch, and have it running on Ubuntu Hardy in ec2 with mod_skypopen, and when I call the echo test with skype it is _beautiful_ and has no lag, and the same is true for my coworker, but when we both dial a mod_conference bridge with skype, we get a 10-12 second lag. CPU usage on the machine is nil. Any ideas?
Do you mean that when you speak, it takes 10-12 seconds before the audio is heard by someone else in the conference?
Correct.
Also, can you try the same exercise with a soft phone like x-lite? I'm curious to know if this happens only on Skype calls or on any calls made to a conference.
Sip is a whole other can of worms that's not working either. When connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo fine, but it won't listen to DTMF numbers to change the menu. When running xopier on linux, I can't hear anything and DTMF doesn't get through. Both those softphones work fine with asterisk, including dtmf. I was punting on getting sip working correctly in freeswitch until I determined whether skypopen solved the conferencing woes I'm facing with MeetMe/asterisk. (It's entirely possible/probably I just have sip misconfigured...I'm just using the default configuration in that regard.) -Clint
-MC
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-- Clint Popetz http://42lines.net Scalable Web Application Development
_______________________________________________ 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 GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
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-- Clint Popetz http://42lines.net Scalable Web Application Development
Sip is a whole other can of worms that's not working either. When connecting with Bria on the mac to the ivr_demo on 5000, I can hear the demo fine, but it won't listen to DTMF numbers to change the menu.
By coincidence, for the last couple of hours I've also been trying to work out why I can't get DTMF on FreeSwitch on EC2 from Bria. Can't even think how to debug it - the Bria diagnostic logs show the dtmf being prepared and sent, but dtmf's don't seem to show on the server if sofia loglevel all 9 is turned on. So if anyone has any ideas on how to work it out, there's two of us that could do with some help :-) In my case I had set things up on a local dev server (with DTMF working, so it's not a pure-Bria problem), then have transferred the same setup to the EC2 server. I'm calling out from behind a NAT, so it could be something related to that maybe. (The differences between the local and remote servers are all related to being on internal vs external networks. Cheers, Fraser
participants (4)
-
Anthony Minessale
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Clint Popetz
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Fraser Redmond
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Michael Collins