[Spce-user] call timeout with no what we suspect is no RTP for the duration

[Matthew Ogden]

Hi All

ACcording to
http://lists.sipwise.com/pipermail/spce-user/2015-January/007929.html

It looks like RTP engine will not cut if one side is not sending RTP.

How can I set it so that if one side of the call is silent for more than 10
minutes, it will cut the call?

We arent using session timers to our asterisk clients because of the
asterisk session timer bug. But here the RTP engine isn't picking up the
failure, because the receiving part put the phone back on his desk and
walked away, unaware the call was still active.

Here is the RTP log to confirm what I'm thinking (very little packets from
RTP[31952] compared to RTP[31950]

Jun 29 08:51:42 spce mediaproxy-ng[1450]:
[0ca94ca13aa1c661112b69ab56fc5ae4 at X.X.X.X] --- side A: RTP[31950] 60122 p,
6852186 b, 0 e; RTCP[31951] 721 p, 76426 b, 0 e; side B: RTP[31952] 1640 p,
184566 b, 0 e; RTCP[31953] 554 p, 48044 b, 0 e

Kind Regards

-- 





*Matthew Ogden*

Management

TenacIT





*Strategic IT Consulting *•* Advanced Networking *• *Virtualisation*

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National Tel: 041 10 10 100 | Cell: 084 205 4445 | Email:
matthew at tenacit.net

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