Clipped audio in recordings

Unanswered Question
Feb 8th, 2010

We have a customer with a CUCM 6 and Genesys contact centre integration. They are using NICE to record inbound calls destined to the agents. What we have found is that the audio in the beginning of each recording is clipped. The file contains approx 1-2 secs of silence and then the rest of the greeting that the agent is saying is heard. So if the agent is actually saying "Welcome to ABC corporation", in the recording the first one or two words are missed and instead there is complete silence. The customer who has dialled in can hear the agent fine. Its just the recording that seems to have a problem. Has anyone come across a similar problem or would happen to have some idea on this? Your help is much appreciated.

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Paolo Bevilacqua Mon, 02/08/2010 - 04:32

You should complain to the recorder maker, not to cisco, since cisco has no part in recording and there is no clipping on a normal conversation.

Nelson Serrao Mon, 02/08/2010 - 13:00

The purpose of my question was to see if anyone else had faced this issue and the the steps that may have been taken to fix it along with some ideas that could possibly assist me in doing root cause analysis. Apologies if it sounded like I was claiming it to be a Cisco issue.

NICE have come back and said that it is a problem on the Cisco end without giving any details which is unacceptable. Hence this post to try and dig deeper into the issue and fix/escalate it with support from the community.

Paolo Bevilacqua Tue, 02/09/2010 - 03:09

The recording vendor is wrong and just doing the old fingerpoinitng game.

Remember, with the span port, the recording system always sees the entire media stream. So, it cannot be Cisco fault.

I have a system by another vendor, and no issues of clipping.

Nelson Serrao Wed, 02/10/2010 - 04:22

I went through the vendors documentation and they are not using the SPAN port configuration for recording.

There is a SIP trunk created to their proxy server which then sends the IP address of the recording server using SIP messages back to CUCM.

I also had a look at http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/6_1_1/ccmfeat/fsmr.html#wp1054230 to get a better understanding of Cisco's BIB feature on the IP Phones.

We setup a packet captue using Wireshark and connected the sniffer to the PC port of the phone. As per Cisco documentation, we were able to see 4 streams. However, in one of the streams to the recording server, it shows a delta of 1000ms and on further analysis, it reports "Wrong sequence number" for RTP packets.

Is this some kind of a delay that gets introduced when using the BIB feature which in turn needs to invoke the phone DSP?

Could it possibly be a bug or something? CUCM ver 6.1.3, phone type 7941 and SCCP load 8.4.2

Copy of capture attached for reference.

Any ideas/suggestions?

Attachment: 
Paolo Bevilacqua Wed, 02/10/2010 - 04:26

So you are using forking recoderding, that is a newer tecnique than span.

Either involve Cisco TAC, or use span that is known to work well and stable.

markbatts Wed, 02/10/2010 - 06:21

Hi,

This might not be of much help but i worked on the first uk deplyoment of the NICE recording solution using the SIP and the BIB capability of CM6.

I hate to say this but we had so many problems that after about two weeks we had to remove it and resort back to standard spanning recording.I think there is still some work to be done on this from cisco and the various recording vendors

Cheers

Mark

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