Cisco Built in Bridge Feature not working while make internal call, Interal Conference call recording its working fine...

[vishal agavane]

Hi,

 

I have a problem when configuring recording with Nexlog Recorder. I use BIB as Call Recording features on CUCM 11.5 instead of SPAN.

I have prepared everything for this method and tried the below scenarios:

Phone A (8861): Built in bridge enable, automatic recording enable

Phone B (8861): Built in bridge enable, automatic recording enable

Phone C (8861): Built in bridge enable, automatic recording enable

1. Making call from phone A to Phone B, Recording not working. I can see in recorder SIP Singnaling is working fine however IP phone not sending any RTP stream to the recorder, when i logged into IP phone web browser i couldn't see any ip (Recorder IP) in remote address of stream-2.

2. Making conference call between phone A, B & C, recorder start recording, I can see Recorder ip in 2nd stream of IP phone web.

I use Wireshark to capture packet from Recording server. In the 1st scenario, I cant see RTP streams are being sent from IP Phone to Recording Server. However in 2nd case, there are RTP steam coming to the recording server.

So, is there any additional configuration in my case?

Thank you..    

[Ayodeji Okanlawon]

Have you tried changing the recording media source to "phone preferred"

Did yo0u also configure the recording profile for the phone?

[vishal agavane]

Thanks for your reply,

Please find your query answer below.

Have you tried changing the recording media source to "phone preferred":- Yes it was already in place

Did yo0u also configure the recording profile for the phone? Yes it was already in place

I have IP phone with "sip88xx.11-5-1-18"firmware loaded.

Your help to understand issue would be appreciated.

[Ayodeji Okanlawon]

Please collect CUCM logs and send over. Use the above link to learn how to use RTMT to collect cucm logs.

Include the calling and called number and time of call

https://www.cisco.com/c/en/us/support/docs/unified-communications/unified-communications-manager-callmanager/200787-How-to-Collect-Traces-for-CUCM-9-x-10-x.html

[vishal agavane]

Hi,

Please find attached file collected file for 10 min interval in which only Calling party (1922) to called party (1015) was in the call, I've tried with other way calling as well.

Thanks,

Vishal 

[Ayodeji Okanlawon]

Hi Vishal,

After looking at the logs the issue is codec related. The original call was completed using opus codec. But when an INVITE was sent to the call recorder, it responded with only g711u,g711a and g729.

+++ Analysis ++++

+++ 200 Ok from call recorder which indicates it only supports, g711u, alaw and g729 +++

00074802.000 |14:04:44.479 |SdlSig |SdlDataInd |wait |SIPUdp(1,100,71,1) |SdlUDPConnection(1,100,10,1) |1,100,10,1.453^10.224.6.42^* |*TraceFlagOverrode
00074802.001 |14:04:44.479 |AppInfo |//SIP/SIPUdp/wait_SdlDataInd: Incoming SIP UDP message size 773 from 10.224.6.42:[5060]:
[9280,NET]
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.224.5.209:5060;branch=z9hG4bK3903d885ceb
From: <sip:1015@10.224.5.209;x-nearend;x-refci=27597150;x-nearendclusterid=StandAloneCluster;x-nearenddevice=SEP00F82C3ED4F0;x-nearendaddr=1015;x-farendrefci=27597149;x-farendclusterid=StandAloneCluster;x-farenddevice=SEPF8A5C5A15324;x-farendaddr=1922>;tag=3254~cecd528f-0d07-4013-8110-9b16ea7a1680-27597158
To: <sip:9999@10.224.6.42>;tag=5822SIPpTag08b417
Call-ID: 7911f380-9e91d84c-2e2-d105e00a@10.224.5.209
CSeq: 101 INVITE
User-Agent: Eventide.
Contact: <sip:9999@10.224.6.42:5060;transport=UDP>
Content-Type: application/sdp
Content-Length: 133

v=0
o=user1 53655765 2353687637 IN IP4 10.224.6.42
s=DESKTOP
c=IN IP4 10.224.6.42
t=0 0
m=audio 6832 RTP/AVP 0 8 9
a=recvonly

+++ Here we see CUCM complaining that it cant insert opus codec into SDP +++

00074800.030 |14:04:44.479 |AppInfo |//SIP/SIPHandler/ccbId=0/scbId=0/insertOPUSParameters: OPUS -- error while adding OPUS codec to the SDP

--
--

+++ Next CUCM tries to invode a xcoder since there is a codec mismatch, but it didnt, hence the call failed +++
00074834.005 |14:04:44.480 |AppInfo |MRM::getXcodeDeviceGivenMrgl GETTING XCODE FROM DEFAULT LIST
00074834.006 |14:04:44.480 |AppInfo |MediaResourceManager::sendAllocationResourceErr - ERROR - no transcoder device configured
00074834.007 |14:04:44.480 |AppInfo |GenAlarm: AlarmName = MediaResourceListExhausted, subFac = CALLMANAGERKeyParam = , severity = 4, AlarmMsg = MediaResourceListName : MRG_List1
MediaResourceType : 2
MediaResourceTypeValue : Transcoder
--

+++ CUCM dropped the call with cause code 47 +++
00074852.001 |14:04:44.481 |AppInfo |SIPTcp - wait_SdlSPISignal: Outgoing SIP TCP message to 10.224.169.3 on port 49307 index 1077
[9282,NET]
BYE sip:11df1f84-4094-48bf-b0d7-a265628cd7b7@10.224.169.3:49307;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.224.5.209:5060;branch=z9hG4bK39230b9ac6b
From: "Call Manager" <sip:10.224.5.209>;tag=3251~cecd528f-0d07-4013-8110-9b16ea7a1680-27597154
To: <sip:1015@10.224.5.209>;tag=00f82c3ed4f0000a605111c8-42c043ba
Date: Fri, 20 Oct 2017 11:04:44 GMT
Call-ID: 7911f380-9e91d84c-2e0-d105e00a@10.224.5.209
User-Agent: Cisco-CUCM11.5
Max-Forwards: 70
CSeq: 102 BYE

Reason: Q.850;cause=47

+++ Suggestions +++

You have two options

1. Change the order of codec preference in your audio codec preference list to G711ulaw so that G711ulaw is used for the call

2. Configure a xcoder that can xocde between opus and G711u/a used by your recorder

[vishal agavane]

Thanks for your detail analysis.

I've seen in CUCM under  telephone configuration option -->"Advertise G.722 and iSAC Codec"setting was "Use System Default" I changed to "Disable" and issue got resolved.

Before changing above setting i could see under IP Phone web interface --- stream-1 coded was 'none" and there were no ip address for Remote Address when there is no communication and whenever we have VOIP communication i could see Codec was "OPUS"and remote address was the ip address of calling party. Please refer attached screenshot for your reference.

Please let me know this is what you were expecting me to do?

Thanks

[Ayodeji Okanlawon · Accepted solution]

I am glad issue is resolved. This was not what I was asking you to do but its working now, so this is fine. 

Don't forget to rate or mark as answered 

[vishal agavane · Accepted solution]

Thanks for your help!!!

[vishal agavane]

Hi, Just last question in this regards.

Whenever BIB recording starts phone announce one beep sound after periodic interval.

Can we stop this beep sound to be notified by end user? If yes then what are the setting we needs to do?

[Ayodeji Okanlawon]

You can disable recording tone on the phone as follows

Product Specific Configuration Layout

recording tone: Disable

[vishal agavane]

It worked...

Thanks for your throughout support.

[larryrevis]

We had this problem too, except only internal calls where not being recorded. This happen after updating to sip88xx.12-0-1 "firmware loaded.. We disabled opus for "call recording only"