[SR-Users] Best practice to figure out one way rtp delay

[Karsten Horsmann]

Hi David,

I have a lot of traces. Today I bought in an cubro Tapping device to
capture all rtp between internet and as second point the b2bua.

But Wireshark is not helpful on all calls, if the corresponding sip
signaling is out of the captures (I have that one the inside).


It's a kind of black box testing to figure out where 386ms delay in one rtp
direction can come. The isp showed me his MOS and jitter on there side and
it was good. But it's an delay in the call 1 direction.

Tommorrow is another great delay hunting day.

Cheers
Karsten


David Villasmil <david.villasmil.work at gmail.com> schrieb am Do., 6. Feb.
2020, 18:42:

> Have you already taken a look at traces?
> Have you checked FS's MOS score, jitter, etc?
>
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Thu, Feb 6, 2020 at 6:43 AM Karsten Horsmann <khorsmann at gmail.com>
> wrote:
>
>>  Hi yuriy,
>>
>> Of course per default Kamailio don't handle rtp. I forgot that I use
>> rtpengine on the "sbc" with Kamailio.
>>
>> And as you all know, if it's sounds bad, then we sip guys get the
>> tickets...
>>
>> So that's an question for your experience in this area.
>>
>> Cheers
>> Karsten
>>
>> Yuriy Gorlichenko <ovoshlook at gmail.com> schrieb am Do., 6. Feb. 2020,
>> 10:27:
>>
>>> Kamailio does not handle RTP.
>>>
>>> But you can use heplify ( or homer)
>>> That will alliw you to collect rtcp in your infrastructure and map it
>>> with the SIP calls you have.
>>>
>>> On Thu, 6 Feb 2020, 09:23 Karsten Horsmann, <khorsmann at gmail.com> wrote:
>>>
>>>> Hi List,
>>>>
>>>> I have a new setup with two Kamailios installations. One serves
>>>> siptrunks from the internet and one is for internal routing.
>>>>
>>>> So far so good. At the end I have an 3rd party b2bua that receives and
>>>> sends calls via the both Kamailios.
>>>>
>>>> For testing purposes I setup an freeswitch with beep and then echo
>>>> application to the caller.
>>>>
>>>> My call flow are like this
>>>>
>>>> Call 1 to kam1 sbc then internal kam2 and b2bua.
>>>> B2bua make then new call no 2 vice versa to PSTN freeswitch with echo.
>>>> After this b2bua bridges the calls together.
>>>>
>>>> This generates for the caller an beep and echo.
>>>>
>>>> The interesting thing is now, that Call 1 gets an hearable delay of 1
>>>> second.
>>>>
>>>> But only in the rtp steam from me to the caller.
>>>>
>>>> The second calls seems equal of timing.
>>>>
>>>> Since is a really new complete setup of hardware and stuff, the
>>>> question of its working before is answered with an no.
>>>>
>>>>
>>>> Now my question to you guys.
>>>>
>>>> How can I get an measurable method to finding the delaying parts (could
>>>> be network, servers, applications etc).
>>>>
>>>> Only capturing on one place don't did the trick for me.
>>>>
>>>> Thanks for your hints
>>>>
>>>> Cheers
>>>> Karsten
>>>> _______________________________________________
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>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
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