[Freeswitch-users] Fwd: Strategies for testing latency

[Kristian Kielhofner]

Darren,

  Relatively anyone can set themselves up as a "Tier X" carrier.  Your
hypothesis is most likely correct; many carriers are running Asterisk
(or who knows what) and ( somewhat unnecessarily) handling media for
billing, topology hiding, etc.  That's why I typically only deal with
carriers I know are Tier 1 LECs.  I feel much better when they own and
control the media gateways and TDM circuits.

  Anyways, a while back I started a project to investigate the actual
media quality provided by carriers.  I found that (in many cases) the
IP path between our endpoint and the remote end (per the media
stream/SDP) would be fine but the quality would still be terrible.  I
needed a tool to measure the quality of the actual audio stream end to
end (as experienced by the actual user).

  I know the ITU and others have devised similar standards and tools
for doing the same thing but I wanted something that could be easily
tweaked and extended for various needs so I created recqual:

http://admin.star2star.com/recqual/RECQUAL-README.txt

  There's some other stuff on google if you'd like to check it out but
it should be fairly easy to adapt recqual to give end to end delay
numbers...

On Tue, Nov 2, 2010 at 1:43 PM, Darren Schreiber <d at d-man.org> wrote:
> Hi folks,
>
> This may be a slightly off-topic FreeSWITCH question, but here goes...
>
> I am having issues trying to figure out the source of latency in circuits we
> LCR to, it seems to be somewhat of a guessing game. I am embarking on
> creating a tool to measure this latency so we can test carriers more rapidly
> when deciding whether to use them or not.
>
> Before I re-invent the wheel, I'd love to know what others do to ensure
> latency is at a minimum on the circuits they buy from carriers OR if there
> are specific things to tweak in FreeSWITCH that I'm unaware of that somehow
> would help this. I can't imagine there are, but maybe I've missed something.
> It is my belief that some of the carriers we are using are behind Asterisk
> boxes that are always taking on media and are adding an encode/decode step
> to the path, resulting in some latency, as I can't find any evidence of late
> packets or high jitter on our box <-> carrier.
>
> For us, latency seems most noticable when we have callers on cell phones
> using a conference bridge. The latency nears 800-1000ms from when a person
> speaks to when they are heard on the other cell phone. I have tried to tweak
> our own conference bridge settings but I really don't think that's going to
> make a difference if my theory of the issue being upstream is correct. VoIP
> <-> VoIP conferencing has none of this issue.
>
> Any experience that folks can offer here is appreciated. Obviously getting a
> direct RTP stream from the carrier doing the conversion to PSTN would be
> best, but it has a tendency to require a lot of minutes or be expensive to
> do yourself.
>
>
> Thanks,
>
> Darren Schreiber
>
> Co-Founder - the 2600hz Project
>
> (415) 886-7900 - www.2600hz.org
>
> Join us in January for FreeSWITCH Training in Australia!
>
> Visit www.voipkb.com for more information
>
>
>
>
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-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com