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One way audio after 30-minute call

We have a CUCM ver9.1, connected to ISP through 2xT1 links (using 2x 2911 Cisco routers), but we know our ISP has an addtional connection via SIP to get PSTN.

My client PSTN connection is like this:

CUCMv9 <--------------------> Cisco 2911s <----------------->ISP equipment <------------------>PSTN

                     h323 T/K                                      2xPRI                                      SIP T/K

My client has observed that some calls (the number is increasing) lose one way audio (most of them outgoing audio) after 30 minutes of the call.

Today, after a pair of 1-hour calls with no problem , I got  another one call, that after 30 minutes it  got no audio from client, but the call was still active and my client was able to hear my voice, so the call got mute with outbound audio only (from client perspective).

Do you have any idea ot a possible solution of this kind of problem ??

I really appreaciate if any of you can help us.


Enrique Villasana


Re: One way audio after 30-minute call

Check with the ISP what they use.

For us it turned out to be deep packet inspections feature on our  Juniper SRX Firewall. Once we disabled these features our problem went  away. We disabled the following:

  • SCCP (Skinny protocol)
  • MGCP (Media Gateway Control Protocol – another PSTN VoIP protocol)
  • H323 (Video based but we disabled it anyway)
  • REAL (Real time streaming protocol, can impact UDP)
  • RTSP (Real time streaming protocol, can impact UDP)

You can also have a look at the following post:

9 posts since
24 Apr 2006

11 May 2011 12:32 PM                             (in response to dsmith16)

I ran into  a similar problem where after exactly 30 minutes the call would drop on  our CME router. It was a problem with the REINVITE between CME and our  SIP provider just like Logan mentioned. Here is what we did to fix it,  you may have some luck with this but might have to tweak it depending on  what your provider is looking for:

voice class sip-profiles 100
request INVITE sip-header Allow-Header modify ", UPDATE" ""
request REINVITE sip-header Allow-Header modify ", UPDATE" ""
response 180 sip-header Allow-Header modify ", UPDATE" ""
response 200 sip-header Allow-Header modify ", UPDATE" ""

voice service voip
  sip-profiles 100

Hope that helps you.

- Jason

Voice CCIE #37771
Community Member

One way audio after 30-minute call

Jason, I am just contacting my ISP, and sending your comments, Thanks for your fast answer, I will let you know their results.

Enrique Villasana

Community Member

One way audio after 30-minute call

Hi everyone !!

After one month of testing, the ISP found one equipment on SIP cloud sending a keepalive which did not receive any acknowledge from its remote peer. After 30 minutes without keepalive remote peer cut audio from customer to PSTN. We are asking more information about the solution.


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