<div dir="ltr">You are right Jeff...Thanks a lot <br><br>Regards<br><br><div class="gmail_quote">On Tue, Dec 16, 2008 at 12:35 AM, Jeff LaCoursiere <span dir="ltr"><<a href="mailto:jeff@jeff.net">jeff@jeff.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>
I'll assume that you suspect that asterisk is adding latency that you<br>
would like to tune. There is no simple variable that will affect latency<br>
as far as I know, but certainly one thing to look at is codec translation.<br>
Make sure your inbound and outbound paths are using the same codec, or<br>
latency will be added for sure.<br>
<br>
You can use tcpdump to measure the latency and the effect of anything you<br>
do to attempt tuning in a rough way - each packet has a timestamp at the<br>
beginning measured in ten thousandths (I think?) of a second. You should<br>
be able to see the RTP packet arrive and then leave again... just subtract<br>
the timestamps for your added latency.<br>
<br>
Cheers,<br>
<div><div></div><div class="Wj3C7c"><br>
j<br>
<br>
On Mon, 15 Dec 2008, michel freiha wrote:<br>
<br>
> Dear Sir,<br>
><br>
> What I'm interested to is to know how much time the rtp packets takes from<br>
> the time it access the asterisk server,to when it'll leave<br>
> Is this function or variable exist anywhere?<br>
><br>
> Regards<br>
> On Mon, Dec 15, 2008 at 10:55 PM, Jeff LaCoursiere <<a href="mailto:jeff@jeff.net">jeff@jeff.net</a>> wrote:<br>
><br>
>><br>
>> No. TTL in the header is about hop traversal. Each IP router that<br>
>> forwards the packet will reduce this number in the live packet until it<br>
>> reaches zero, when it will be dropped. I believe this is to eliminate<br>
>> route loops creating packet storms.<br>
>><br>
>> FWIW this is how traceroute works - it sends out packets with continually<br>
>> increasing TTLs and the router that drops the packet will send back a<br>
>> notification, so you can "trace" each hop...<br>
>><br>
>> What is it you are trying to do or measure?<br>
>><br>
>> j<br>
>><br>
>> On Mon, 15 Dec 2008, michel freiha wrote:<br>
>><br>
>>> Dear Sir,<br>
>>><br>
>>> There is no relation between TTL and the latency on asterisk server?<br>
>>><br>
>>> Regards<br>
>>><br>
>>> On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere <<a href="mailto:jeff@jeff.net">jeff@jeff.net</a>><br>
>> wrote:<br>
>>><br>
>>>><br>
>>>> TTL is part of the UDP header (Time To Live). It isn't really about the<br>
>>>> voice at all.<br>
>>>><br>
>>>> Length 345 is the number of bytes in the packet.<br>
>>>><br>
>>>> j<br>
>>>><br>
>>>> On Mon, 15 Dec 2008, michel freiha wrote:<br>
>>>><br>
>>>>> *Dear All,<br>
>>>>> I run the below tcp dump on my asterisk server<br>
>>>>><br>
>>>>> tcpdump -i eth0 -n -s0 -v udp port 5060<br>
>>>>><br>
>>>>> I got the following result<br>
>>>>><br>
>>>>> 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF],<br>
>> proto<br>
>>>> 17,<br>
>>>>> length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345<br>
>>>>><br>
>>>>> What i need to know please what TTL means specifically and what is the<br>
>>>> best<br>
>>>>> value og TTL and what is the lengh vale mean<br>
>>>>><br>
>>>>> Regards*<br>
>>>>><br>
>>>><br>
>>>> _______________________________________________<br>
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>>><br>
>><br>
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><br>
<br>
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