[OpenSIPS-Users] Tearing down the RTPProxy session for the current call
Schoolhouse Filing
schoolhouse at a2es.co.uk
Thu Nov 9 04:15:32 EST 2017
Hi Razvan,
Thankyou for clarifying this, I was a little confused because the
Opensips generated scripts that use rtpproxy for NAT do not call the
rtpproxy_unforce() function.
I will add a call to trpproxy_unforce() in my route that handles BYEs
and in the missed call (failure) routes.
Many Thanks,
Adrian.
On 08/11/17 10:47, Răzvan Crainea wrote:
> Hi, Adrian!
>
> Yes, you should be calling rtpproxy_unforce() when a BYE is received.
> You are right, if you do that, you won't be seeing too many "RTP
> sessions ended due to media timeout".
>
> Best regardsm
> Răzvan Crainea
> OpenSIPS Developer
> www.opensips-solutions.com
> On 11/07/2017 08:59 PM, Adrian Fretwell wrote:
>>
>> Hello,
>>
>> I was just looking at some statistics from my rtpproxy today and
>> noticed that "RTP sessions created", "RTP sessions destroyed" AND
>> "RTP sessions ended due to media timeout" all showed the same number.
>>
>> Although I am aware that the function rtpproxy_unforce() is
>> available, in example routing scripts, I have never seen it used.
>>
>> Should I be calling rtpproxy_unforce() in my routing script at the
>> end of a call?
>>
>> My assumption is that if I do, then the "RTP sessions ended due to
>> media timeout" will be much lower.
>>
>> As always any advice will be gratefully received.
>>
>> Kind regards,
>>
>> Adrian Fretwell
>>
>>
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>
>
>
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