[asterisk-users] issue with high latency

[Jason Aarons (US)]

Jitter is what your describing, it's a bad thing.
http://en.wikipedia.org/wiki/Jitter

While VoIP may work (third party > 128ms echo cancellers, etc) most
support organization won't go outside ITU-T G.114 recommendations.

I've done Cisco 7940 phones deployed in the Gulf of Mexico on a oil
platform using Callmanager based in US in 2003. The company controlled
the satellite and prioritized voice, ping was 600ms. Worked well except
the local calls were to Venezula which was too expensive per minute from
US.  Two phones ran up more than $1000US in 30 days.



-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tom Moore
Sent: Tuesday, July 22, 2008 11:40 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] issue with high latency

Not true.
Voip is done over satellite every day and those ping times are at least
540
and upwards of in the 700's depending on the technology used.
The key here is keeping the latency stable.
If the packet flow fluctuates too much in latency this is when a problem
arises.

Tom

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Steven
Howes
Sent: Tuesday, July 22, 2008 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] issue with high latency


On 22 Jul 2008, at 14:36, Nhadie wrote:
> Pinging my.sipserver.com [202.203.204.205] with 32 bytes of data:
>
> Reply from 202.203.204.205: bytes=32 time=250ms TTL=56
> Reply from 202.203.204.205: bytes=32 time=250ms TTL=56
> Reply from 202.203.204.205: bytes=32 time=651ms TTL=56

Never going to work with that latency. I would say anything over 150  
is probably pushing it.

S

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