[Freeswitch-users] rtp-timer-name variable

[Anthony Minessale]

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 at 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 at 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 at 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 at 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.html
>>>> ).
>>>>
>>>> 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 at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
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>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
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>>>
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>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Thanks and Regards,
>> Ashwin Jain
>> Director (Engineering)
>> MetroGuild
>>
>
>
>
> --
> Thanks and Regards,
> Ashwin Jain
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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