[Freeswitch-users] Media delay when calling FS from a WebRTC client

Vincent Xia gmangudai at gmail.com
Thu Mar 13 04:08:22 MSK 2014


as i observed the delay is caused by the DTLS handshake and the 2.5 sec
delay in answering is a remedy for that, otherwise you'd hear nothing when
the other side is already talking.


2014-03-13 6:09 GMT+08:00 Oleg Stolyar <olegstolyar at gmail.com>:

> One clarification:
>
> I understand that the delay may be caused by handshaking and testing
> different ICE candidate pairs but it seems to happen even when the correct
> candidate from the WebRTC client (the srflx one) has the highest priority
> in the SDP.
>
>
> On Wed, Mar 12, 2014 at 2:57 PM, Oleg Stolyar <olegstolyar at gmail.com>wrote:
>
>> Hi guys,
>>
>> does anyone know what is causing media delay after the call is connected
>> in this case?
>>
>> Usually delays in WebRTC are caused by ICE candidate gathering and I am
>> seeing that on my client (JsSip demo).  However there is another delay in
>> getting media after the call is connected.
>>
>> I found an old Jira ticket where Anthony mentioned that on FS demo
>> servers there is a 2.5 sec delay in answering but I could not find info on
>> what is causing this delay.
>>
>> It seems to me that FS does not need to gather ICE candidates since it
>> can (and does) take the only candidate from the ext-rtp-ip variable.
>>
>> So, why the delay?
>>
>> Thank you
>> Oleg
>>
>
>
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