[Freeswitch-users] RTP keep alive

[Steven Ayre]

In media, there's only the possibility of detecting absence of it. Not sure
if RTCP would contain something, but FS doesn't support support it and
neither do all endpoints.

The alternative (probably preferred) method is session invites, which checks
the call is still up through the SIP signalling.

-Steve



On 9 February 2011 17:01, Sam <u2nsam at gmail.com> wrote:

> There is no vad , is there something like rtp keep live after the increase
> of rtp timout .
> Yes the RTP timeout helps.
>
> Regds
> Sam
>
>
> On Wed, Feb 9, 2011 at 8:03 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> That'll do it.
>>
>> VAD can mean no RTP is transmitted legitimately though, so be careful.
>> Especially when on hold when the phone might stop sending anything (I see
>> you've already increased that time).
>>
>> -Steve
>>
>>
>> On 9 February 2011 08:49, Sam <u2nsam at gmail.com> wrote:
>>
>>>  I have this in the settings,
>>>
>>>     <param name="rtp-timeout-sec" value="3000"/>
>>>     <param name="rtp-hold-timeout-sec" value="18000"/>
>>>
>>> Any thing i need more regarding that ?
>>>
>>> Regards
>>> Sam
>>>
>>>
>>> On Wed, Feb 9, 2011 at 1:35 PM, Sam <u2nsam at gmail.com> wrote:
>>>
>>>> Hello,
>>>>
>>>> We have a situation wherein we need to keep alive RTP ,is there any
>>>> parameter to do that,
>>>> because when someone is on long conversation and not talking fro brief
>>>> duration the call
>>>> disconnects .
>>>>
>>>> Regards
>>>>  Sam
>>>>
>>>
>>>
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