SIP Calls Drop off at 5 Minutes when you Mute them.

[stownsend]

When we call a conferencing service an use the phone to Mute our end (instead of the Conference line's mute feature) the Call will get dropped between 5-6 minutes.

The Provider says that since we stop transmitting all together, the Upstream carriers (yes more than one of them) drop our calls.

On the VoIP Tab of the LAN 1 properties (we are only using LAN1)
I have RTP keepalives Settings as Follows:
scope set to RTP
Timeout 10 Sec
Initial Keepalives Enabled

The only other thing that I can think of is the settings on the Firewall. It is an Cisco ASA 5510.

timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00
timeout sip-provisional-media 0:02:00 uauth 0:05:00 absolute

I can try increasing them to see if that increases the time to drop.

Thanks,
Scott<-

 

[CptIPO]

Why would not use the Conference service's mute function.

Yes we have VoIP in Cape Town

 

[stownsend]

Pressing one button that works with any Conferencing service, vs two or three that are different for a few different services?

I know, its a User Training issue. Though the are used to doing it for so long, and now the system does not work like it did before and its hard to re-train them.

That and I've only been able to get it to fail with 800 type numbers and Conferencing Services. I have not been able to get it to fail with non-800 type numbers, and have not been able to call an 800 type number and have them place me on hold for 6 minutes.

Seems odd that it would only be limited to a conferencing service.

Scott<-

 

[Bas1234]

Tried to enable RTP/RTCP (RTP Keepalives) on the LAN 1 or 2 > Voip Tab...depending where you're SIP provider is on. Set the time to 4 or 5 minutes.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[stownsend]

They have been Enabled from the get go:

Scope: RTP
Initial Keepalives: Enable
Periodic Timeout: 10 (Range = 0 to 180 seconds)


You mentioned 4-5 Minutes, though the Range is only 0-3 Minutes. System, LAN1, VoIP, all the way at the bottom. Is this the Wrong place?

I'm on IPO500v1 R8.0(44)

Thanks

 

[Bas1234]

Indeed seconds, tried RTCP/RTP instead of RTP only?

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[stownsend]

From my understanding RTCP is just the statistics that are reported back to get Call Quality info. Many systems don't use it.

I have not tried, though my Provider said it would not make a difference.

Thank,
Scott<-

 

[Bas1234]

Yes thats tru, but if the provider sends RTCP and the IPO does not answers it the the call will be lost.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________