Lag in voice path , once agent answers the inbound call

[manjunathc24]

Hi , We are coming across the issue with delay in audio path being heard by customer. Though the agent begin his/her welcome script after the call is being answered , it was never heard by customer . even from voice recording also their is no welcome greet captured . Their is approx delay of 3-4 sec.

Platform is on HCS 10.0.1 

Same issue was posted earlier , curious to know if anyone have fix for this  issue ?

https://supportforums.cisco.com/discussion/12371881/agent-answer-audio-delay

Any thoughts , if its the coz;

+DNS ( in remote location)

+ CVP (TcpAckFrequency) 

Cheers, MGC

[Elliot Dierksen]

What kind of PSTN connection are you using? Is it possible to use something like H.323 fast start or SIP early offer? This sounds like a delay in capability exchange and audio path establishment between the outside caller and the agent phone. I have seen this caused by delays in the signaling path, but it is hard to say exactly without more detailed information.

[manjunathc24]

Apologies for the late reply , Elliot!

It WAN link were we receive the calls from customer, we have no choice to opt for H323.

We realize that resultant delay is cumulative of delay added from customer IVR platform before transferring the call to our platform.

Can you shed more light towards "sip early" , on how to achieve this ?

Thanks

[manjunathc24]

My curiosity is due to fact that , their is delay in establishing the RTP btw customer/agent ., thought the call is already conected.

ie: By the time agents starts welcome announcement to the caller , caller can hear the agent or converse with the agent only after initial welcome announcement. Did anyone come across this issue ??

So, considering this case , guessing if early offer can fix my issue . Any thoughts would be appreciated.

[Elliot Dierksen]

The call is connected to the IVR playing a menu, hold music, etc. When the call is transferred to an agent, that RTP stream is torn down and a new one is built between the caller and the agent who receives the call. That is why I asked about how the callers enter your network. Depending on what type of connection that is, that dictates what you can do to attempt to make the establishment of the new audio stream happen more quickly.

[manjunathc24]

Post IVR treatment from Avaya IVR ( in EMEA region)=>MPLS=>Cisco  platform (in singapore CUBE), calls will be transferred to AGENT on cisco platform.

We are on SIP.

Can you shed more light towards "sip early" , on how to achieve this ? ( is it by invoking MTP ).

Thanks

[manjunathc24]

got this interesting thread.

http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube_sip/configuration/15-mt/cube-sip-15-mt-book/do-to-eo.pdf

[Elliot Dierksen]

That document would certainly be a good read. Assuming you can make changes to the CUBE, you could add the following line to the dial peer(s) that face the CUCM that connects to the agents.

 voice-class sip early-offer forced

[Chavdar_Baramov]

Got similar issue - check your agent CPU/memory load when he answers the call. In my case it was CPU spikes that caused the initial delay. Had to upgrade the workstations to sort it out.