[Freeswitch-users] Session recording performance

[Stanislav Sinyagin]

but the interesting thing is, that it's the device that originates the
SIP call is producing a significantly higher jitter than the one
receiving the call.

The originating command looks like this here, with a SIP address as
destination, and a number in local XML dialplan as transfer
application:
https://freeswitch.org/confluence/display/FREESWITCH/Originate+Example



On Thu, Sep 21, 2017 at 7:46 AM, Stanislav Sinyagin <ssinyagin at gmail.com> wrote:
> with jitterbuffer, it now works as expected. So, the sending party was
> generating extra jitter.
>
> I am playing the same audio file in both directions, and then compare
> the recorded audio with the original by Sevana AQuA software. This
> allows me to detect packet loss in a heterogeneous environment where
> the voice goes through RTP proxies and TDM.
>
>
>
>
> On Thu, Sep 21, 2017 at 3:43 AM, jungle Boogie <jungleboogie0 at gmail.com> wrote:
>> What occurs if you send identical audio for the same duration? IOW, play the
>> sound file for each call.
>>
>>
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