Jssip turn authentication fail

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AD Rehman

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Jul 22, 2021, 4:45:57 AMJul 22
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Hello friends, we managed to make phone working for us but now our call get dropped after 30secs or so... We have coturn as our relay server.

On turn console we can see some authentication error at start of call.

Can some advice please how to create that block of code which generate turn user and password?


Thanks

Iñaki Baz Castillo

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Jul 22, 2021, 4:52:33 AMJul 22
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You are assuming this is related to TURN and credentials but I don’t know why you assume that. Given it’s ~30 seconds I’d say ACK after the 200 response to the INVITE is not reaching the target.

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Iñaki Baz Castillo
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AD Rehman

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Jul 22, 2021, 5:11:27 AMJul 22
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You are correct... Without turn no voice with turn voice is there.

But end result is same call drops..


Please advise.What and where to fix? 

If required I can share rtc internal output with error

AD Rehman

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Jul 22, 2021, 6:17:54 AMJul 22
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You are correct... Without turn no voice with turn voice is there.

But end result is same call drops..


Please advise.What and where to fix? 

If required I can share rtc internal output with error

Iñaki Baz Castillo

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Jul 22, 2021, 6:42:48 AMJul 22
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On Thu, 22 Jul 2021 at 11:11, AD Rehman <adre...@gmail.com> wrote:
> But end result is same call drops..
>
> Please advise.What and where to fix?
>
> If required I can share rtc internal output with error

I already told it:

> Given it’s ~30 seconds I’d say ACK after the 200 response to the INVITE is not reaching the target.

This is a common problem due to bad configuration in SIP
proxies/servers. There is plenty of threads and issues like this even
in this mailing list.

AD Rehman

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Jul 22, 2021, 7:30:19 AMJul 22
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No tried but failed to find any clues...

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Iñaki Baz Castillo

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Jul 22, 2021, 8:03:53 AMJul 22
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At some point you have to get SIP traces or logs in your SIP proxy/server and see if the ACK reaches the target. This is how SIP protocol should work. I cannot help much more.

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