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

[Oleg Stolyar]

Thank you Vincent!


On Wed, Mar 12, 2014 at 6:08 PM, Vincent Xia <gmangudai at gmail.com> wrote:

> 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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>
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