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Call drop after 30 seconds of perfect silence!


Jan Boguslawski

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Hi @all Unified Communications friends :)

 

I like to spotlight you on an issue with OCS R1 and R2 regarding a call drop after exact 30 seconds. This happens in some integration scenarios with IP-pbx, some VoIP-Gateways and maybe also with VoIP-providers.

 

It seems to be related how OCS components / roles check RTP streams on silence (both directions). One example of that can be found here with Cisco Unity.

 

In pbxnsip you can face this problem e.g. when you put an external caller on hold in Office Communicator or R2-Attendant Console. Typically Mediation Server will send a bye after the 30 sek.

 

Drago (in the forum discussion about the Unity issue) has an interesting explanation:

If you trace the Medation stack, you'll see that upon placing the call on hold, MOC devices will triger ProxySDP request, which in turm makes Mediation server to keep the channel alive.

 

If SDP is not trigered, after 30 sec., w/o RTP, Mediation will send GOODBY and close the channel.

Snom with their native OCS Edition phones also faced that problem in a similar way. But they have find a workaround J They are sending short none audible audio during the 30s.

 

Maybe this workaround can be featured at an special pbxnsip trunksetting too? Or maybe a OCS native solution, which e.g. is done by AudioCodes Gateway's for Microsoft UC (sorry I dont know the details atm. but I can provide ACsyslog traces, where you might see what is happening if you put someone onhold in a direct OCS Mediation <-> AC Gateway scenario. I think they dont send none audible audio. :)

Regards,

Jan

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Hi Jag,

 

We are using Version: 4.0.0.3344 (Linux) and still experiencing problems with call drop outs after 30 seconds, we are using a Matrix GSM gateway registered to pbxnsip to make outbound calls to mobile phones , after exact 30 seconds call to mobile phones are terminated, calls to a landline works fine. any ideas what it could be the problem?

 

When you use the gsm gateway without register on pbxnsip works fine too

 

Any help would be appreciated

 

 

 

Thank You

 

Thiago

 

 

 

This will be added to V4 which is due out towards the end of the year.
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