Audio Delay with 924E

[ndorr]

We recently started to deploy the Adtran 924E 3rd Gen. When we build a hunt group, there is a 2-3 second delay from the time the called party speaks and the calling party can hear.

We have looked at other posts where the "double re-invite" had been removed, we tried and it did not work.

We need help.

Here is our trunk config:

voice trunk T01 type sip

  caller-id-override emergency-outbound 8153792131

  match dnis "NXX-XXXX" substitute "1-815-NXX-XXXX"

  sip-server primary x.x.x.x (removed the SBC's for privacy reasons)

  sip-server secondary x.x.x.x (removed the SBC's for privacy reasons)

  domain "stratus.net"

  rtp dtmf-relay offer nte 101

  no prefer double-reinvite

Here is our voice user and voice ring-group configs:

voice grouped-trunk SIP

  trunk T01

  accept $ cost 0

!

!

voice user 1

  connect fxs 0/1

  no cos

  password encrypted "4642275d03ad2627cea13e3b8fe320bf9b29"

  caller-id-override external-number 8153792131

  sip-authentication password encrypted "2c28de02811752912fc8b57565b9b766ed74"

  t38

!

!

!

voice user 2

  connect fxs 0/2

  no cos

  password encrypted "42464ccdbdf56addca4edf6f3a94425d4e9e"

  caller-id-override external-number 8153792131

  sip-identity 8153792132 T01

  sip-authentication password encrypted "2a2e91b25728c5115d45ab133f07d353344f"

  t38

!

!

!      

voice user 3

  connect fxs 0/3

  no cos

  password encrypted "42464ccdbdf56addca4edf6f3a94425d4e9e"

  caller-id-override external-number 8153792131

  sip-identity 8153792133 T01

  sip-authentication password encrypted "2125a983579bb686fa2262b5796c652c338a"

  t38

!

!

voice ring-group 8153792131

  type linear

  num-rings 4

  member 1

  login-member 1

  coverage  internal 2000

  coverage  internal 2001

  no prefix

!

!

voice ring-group 2000

  type linear

  num-rings 4

  member 2

  login-member 2

  no prefix

!

!

voice ring-group 2001

  type linear

  num-rings 4

  member 3

  login-member 3

  no prefix

!

!

[Anonymous]

Nicolas,

Can you do the following command and give output back to me:

show run voice grouped-trunk

Also can you enter the following debug commands and place an inbound call to the ring group.

debug sip stack messages

Debug voice verbose

debug int fxs

then get the output and put in text file and attach it to your reply.

Thanks,

Mark

[ndorr]

Mark,

Thank you for your response, I have run each command as you requested, attached is the result.

And here is the output from the show run voice grouped-trunk:

show run voice grouped-trunk

voice grouped-trunk SIP

  trunk T01

  accept $ cost 0

[Anonymous]

Nicholas,

Can you do a "show run" and post your config as an attachment. Make sure to remove all passwords.

When exactly are you experiencing the 3 second delay? Is it when your talking? Does it happen in your outbound/inbound talkpath or both directions?

Was FXS 0/1 offhook when you did that test call? I noticed the call went to FXS 0/2 which is second on your hunt group.

Thanks,

Mark

[ndorr]

When exactly are you experiencing the 3 second delay? when the call is first picked up

Is it when your talking? No

Does it happen in your outbound/inbound talkpath or both directions? If I am reading your question correctly it is happening in the outbound talkpath. I say that because when I dial the 3121 number (which is in the ring-group) the person that picks up talks right away, and only after 2-3 seconds I can hear her. It seems like the RTP is late to the party

Was FXS 0/1 offhook when you did that test call? (Sorry I did not check that)

I noticed the call went to FXS 0/2 which is second on your hunt group.

[Anonymous]

Try making this change and then enter the same debug and place an inbound call and post results again.

voice grouped-trunk SIP

reject 8153792131

end

Debug Commands:

debug sip stack messages

Debug voice verbose

debug int fxs

-Mark

[ndorr]

That fixed it,

Now the question is, why did it fix it?

[Anonymous]

Whoot Whoot!  😃

When I went through your original debug of call I noticed that when the call came inbound, the call was hitting the hunt group but then the AOS switchboard was sending a reinvite back up to the softswitch to the hunt group, so I just wanted to block the hunt group number on the outbound truck to prevent that. That will keep the hunt group local to the TA 900 and send it back up to the softswitch. Hope that makes sense.

Good way to view the logs is to use notepad++ and then you can highlight something and right click and go to style token and assign it a color, up to 5 colors. Then it will highlight it all the way in in document. I typically highlight the Call-id and phone numbers and lets you follow the call path for the whole SIP dialogue. 

Glad you got it working! Have a great day.

-Mark