[Freeswitch-users] Skypiax latency
Peter P GMX
Prometheus001 at gmx.net
Thu Feb 11 11:53:34 PST 2010
Hello Hristo,
I am using xm-curl, so xml is processed. But your hints really helped me
to tie the problem down.
As described, my call scenario is as follows:
SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
When I call from Skype to SIP and
* let the SIP phone ringing for 1 sec, I receive a ~1,5 sec delay
from SIP to Skype
* let the SIP phone ringing for 2 sec, I receive a ~2,5 sec delay
from SIP to Skype
* let the SIP phone ringing for 3 sec, I receive a ~3,5 sec delay
from SIP to Skype
So - the longer the SIP phone rings - the longer is the delay.
I also tested the other direction:
If I call from SIP to Skype the behaviour is exactly the same: The
longer the Skype phone rings - the longer is the delay from SIP to Skype.
This may be somehow related to MODSKYPIAX-29, but is a different
scenario. Should I open a new Jira for this or should I attach my
findings to MODSKYPIAX-29?
Best regards
Peter
Hristo Trendev schrieb:
> Hi Peter,
> Take a look at http://jira.freeswitch.org/browse/MODSKYPIAX-29 and
> http://jira.freeswitch.org/browse/MODLANG-130. I haven't tested the
> trick in MODLANG-130 with skypiax yet, but if you use script to handle
> the calls this may help.
>
> BR,
> Hristo
>
>
> On 2/10/10, Peter P GMX <Prometheus001 at gmx.net> wrote:
>
>> Hello,
>>
>> I have a problem with latency and mod_skypiax
>>
>> Skype=>SIP is always fine (~0.3sec)
>> SIP => Skype is always bad (~2-4 sec)
>> I would expect that latency in both directions should be the same.
>>
>> Anybody has discovered this before and has a solution?
>>
>> The scenario is as follows:
>>
>> SIP-UA => FreeswitchServer=>Freeswitch/SkypIaxServer=>Skype
>> Both freeswitch servers are in the same LAN, so latency should be low.
>>
>> Best regards
>> Peter
>>
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>
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