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@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@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@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@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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