Hi, I was facing lot of audio quality issues and turning off sip jitter buffer didn't solved anything. I set rtp-timer-name = none (it was set to soft earlier) and it seemed to solved the problem ( http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.ht... ). Can someone explain what was the issue and how it got fixed? I wasn't able to find any documentation related to "rtp-timer-name". -- Thanks and Regards, Ashwin Jain
You are basically turning off the feature that allows the remote end to stop sending audio but have FS still process the stream. The timer will allow the channel to read at the specified interval even when the other side stops sending audio. If the other end has a clock sync issue with FS or FS does not have a reliable timing source, it could cause your problem. If you happen to have a sipura or linksys, check the wiki for specific ways to mitigate some issues. On Mon, Dec 2, 2013 at 6:06 AM, Ashwin Jain <ashwinrkjain@gmail.com> wrote:
Hi,
I was facing lot of audio quality issues and turning off sip jitter buffer didn't solved anything. I set rtp-timer-name = none (it was set to soft earlier) and it seemed to solved the problem ( http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.ht... ).
Can someone explain what was the issue and how it got fixed? I wasn't able to find any documentation related to "rtp-timer-name".
-- Thanks and Regards, Ashwin Jain
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
If FS has stopped reading audio (I have set it to none), then how that is improving the latency issue? We have EC2 deployment of FS. We are using freeswitch to bridge the incoming call from voip.ms and terminating it using les.net. On Tue, Dec 3, 2013 at 12:53 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You are basically turning off the feature that allows the remote end to stop sending audio but have FS still process the stream. The timer will allow the channel to read at the specified interval even when the other side stops sending audio.
If the other end has a clock sync issue with FS or FS does not have a reliable timing source, it could cause your problem.
If you happen to have a sipura or linksys, check the wiki for specific ways to mitigate some issues.
On Mon, Dec 2, 2013 at 6:06 AM, Ashwin Jain <ashwinrkjain@gmail.com>wrote:
Hi,
I was facing lot of audio quality issues and turning off sip jitter buffer didn't solved anything. I set rtp-timer-name = none (it was set to soft earlier) and it seemed to solved the problem ( http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.ht... ).
Can someone explain what was the issue and how it got fixed? I wasn't able to find any documentation related to "rtp-timer-name".
-- Thanks and Regards, Ashwin Jain
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Thanks and Regards, Ashwin Jain Director (Engineering) MetroGuild
Also, even after setting the rtp-timer-name to none, we are facing 1-2 sec latency (earlier it was 8-10 secs). We also have another server running asterisk, so to check if it's origination provider or termination provider connectivity issue, we routed a call via asterisk. We didn't faced any latency issue with that. Is their something else I should look for? On Tue, Dec 3, 2013 at 2:07 AM, Ashwin Jain <ashwinrkjain@gmail.com> wrote:
If FS has stopped reading audio (I have set it to none), then how that is improving the latency issue? We have EC2 deployment of FS. We are using freeswitch to bridge the incoming call from voip.ms and terminating it using les.net.
On Tue, Dec 3, 2013 at 12:53 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You are basically turning off the feature that allows the remote end to stop sending audio but have FS still process the stream. The timer will allow the channel to read at the specified interval even when the other side stops sending audio.
If the other end has a clock sync issue with FS or FS does not have a reliable timing source, it could cause your problem.
If you happen to have a sipura or linksys, check the wiki for specific ways to mitigate some issues.
On Mon, Dec 2, 2013 at 6:06 AM, Ashwin Jain <ashwinrkjain@gmail.com>wrote:
Hi,
I was facing lot of audio quality issues and turning off sip jitter buffer didn't solved anything. I set rtp-timer-name = none (it was set to soft earlier) and it seemed to solved the problem ( http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.ht... ).
Can someone explain what was the issue and how it got fixed? I wasn't able to find any documentation related to "rtp-timer-name".
-- Thanks and Regards, Ashwin Jain
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Thanks and Regards, Ashwin Jain Director (Engineering) MetroGuild
-- Thanks and Regards, Ashwin Jain
As I already explained. When timer-name set to none, it does blocking reads on the audio vs timed ones. If the other end has a skew in clock freq with the FS then it will timeout erroneously. Having blocking reads ensures you only read audio when there is actually some there. Basically this timer-name set to none makes FS behave just like asterisk does by default. Keep in mind we do not support virtual servers beyond friendly advice because it quickly turns into time-consuming project to help people configure their virtual envs. The basic idea is, unless you are using the high yield cpu version of ec2, you can have timing issues at any time because the EC2 will steal the cpu at will. You should probably get some pcaps and look at the deltas and timestamps on each side of the call. On Mon, Dec 2, 2013 at 2:43 PM, Ashwin Jain <ashwinrkjain@gmail.com> wrote:
Also, even after setting the rtp-timer-name to none, we are facing 1-2 sec latency (earlier it was 8-10 secs).
We also have another server running asterisk, so to check if it's origination provider or termination provider connectivity issue, we routed a call via asterisk. We didn't faced any latency issue with that.
Is their something else I should look for?
On Tue, Dec 3, 2013 at 2:07 AM, Ashwin Jain <ashwinrkjain@gmail.com>wrote:
If FS has stopped reading audio (I have set it to none), then how that is improving the latency issue? We have EC2 deployment of FS. We are using freeswitch to bridge the incoming call from voip.ms and terminating it using les.net.
On Tue, Dec 3, 2013 at 12:53 AM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
You are basically turning off the feature that allows the remote end to stop sending audio but have FS still process the stream. The timer will allow the channel to read at the specified interval even when the other side stops sending audio.
If the other end has a clock sync issue with FS or FS does not have a reliable timing source, it could cause your problem.
If you happen to have a sipura or linksys, check the wiki for specific ways to mitigate some issues.
On Mon, Dec 2, 2013 at 6:06 AM, Ashwin Jain <ashwinrkjain@gmail.com>wrote:
Hi,
I was facing lot of audio quality issues and turning off sip jitter buffer didn't solved anything. I set rtp-timer-name = none (it was set to soft earlier) and it seemed to solved the problem ( http://lists.freeswitch.org/pipermail/freeswitch-users/2011-August/075616.ht... ).
Can someone explain what was the issue and how it got fixed? I wasn't able to find any documentation related to "rtp-timer-name".
-- Thanks and Regards, Ashwin Jain
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
-- Thanks and Regards, Ashwin Jain Director (Engineering) MetroGuild
-- Thanks and Regards, Ashwin Jain
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
participants (2)
-
Anthony Minessale
-
Ashwin Jain