[Asterisk-Users] g729 and latency measures

[Rich Adamson]

Yes, 300ms seems very high if there is no satellite link involved.

g729 should be just fine if that's what you're stuck with.


Erick Perez wrote:
> Thanks Rich, but i'm only allowed to use g729.
> you said that some folks run high latency connections, but is 300ms
> high in my setup?
> 
> On 3/19/06, Rich Adamson <radamson at routers.com> wrote:
>> Erick Perez wrote:
>>> Hi, we have set up a small project in a school the following way:
>>> SITE_A(4 port analog to ip
>>> g729)------ADSL_ISP1-------ISP2--------Asterisk-----PSTN
>>> Site A has 1 Megabit of bandwith (up 512kilobit down 1 megabit)
>>> The asterisk box gets internet service via a wireless antenna. 1 Mbit
>>> of up/down bandwith
>>>
>>> Comments:
>>> So far, this means that I will need licenses for the 729.
>>> asterisk only supports 20ms sampling on g729 so 4 channels will need
>>> 96 kilobits at 20ms sampling (or is it kilobytes??) for the internet
>>> bandwith.
>>> i cannot use CRTP because i cant be sure if the ISP's routers are CRTP aware.
>>> Installing ADSL from ISP1 on the asterisk place will give a clear advantage
>>>
>>> Please correct any of my prior statements if wrong.
>>>
>>> should I maintain packet latency below 300ms or 150ms?
>> The objective should be to keep latency as low as possible, however some
>> folks do run asterisk via satellite which as a very lengthy latency.
>>
>>> How can I measure this latency all the way to the asterisk?
>> Several ways depending on how accurate a measurement you want. A simple
>> ping would give a starting point. A much more expensive way is to use
>> VoIP analysis software to measure it, but be prepared to spend at least
>> $1,500 (US) to do that.
>>
>>> Should I ping from SITE_A to the asterisk box with 8k packets?
>> If you want to emulate a sip/iax packet, use a packet size of about 200
>> bytes.
>>
>>> If I can't install ADSL for the moment, will the above setup work?
>> Probably a bigger issue to address relates to what "other" traffic might
>> be passing across the dsl and/or wireless channel that might be
>> consuming bandwidth and impacting the rtp packets.  Broadcasts
>> originating from devices outside your control (other isp users), hackers
>> attempting to access your ip addresses (at both ends), data traffic
>> between your two endpoints, etc, are just some thoughts of items using a
>> portion of the bandwidth available.
>>
>> Might also think about jitter (eg, variations in latency) and what that
>> might do to your end to end communications.
>>
>> There are other low bandwidth codecs available that could be used
>> instead of g729. Some include ilbc, g726, gsm, etc. Each consumes
>> different bandwidths, and each provide a slightly different quality of
>> audio. See the wiki for more detail on what each consumes for bandwidth
>> on the wire.
>>
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
> 
> --
> 
> -------------------------------------------
> Erick Perez
> Linux User 376588
> http://counter.li.org/  (Get counted!!!)
> Panama, Republic of Panama
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users