[cisco-voip] BIB Configuration Help

Stephen Welsh stephen.welsh at unifiedfx.com
Thu Mar 28 11:10:03 EDT 2013


Have you looked at stream 2 on both phones web page to confirm the RTP source/destination are consistent with the Wireshark trace, is it possible you are looking at the wrong RTP stream?

Our software (PhoneView) has been used on most UCM versions from 6.0+ with the phones BIB with our 'Remote Audio' feature, never had a single issue. Is the recording system handing the 'recording call(s)' over the SIP trunk correctly, it may be responding with an incorrect media description?

I'm curious which product this is?
If you are creating your own might be worth asking on the Cisco developer community (http://developer.cisco.com/web/sip/community).

Thanks

Stephen Welsh
CTO
http://www.unifiedfx.com

[cid:CBBF0493-235C-43D2-A874-9FB3D95AF598 at b2.unifiedfx.com]

On 28 Mar 2013, at 13:55, Scott Voll <svoll.voip at gmail.com<mailto:svoll.voip at gmail.com>>
 wrote:

Someone else can chime in at any point, but the only way I can see that you would get RTP packets from CM, is if it's running through a MTP, Confernce, or transcoder based on a CM server.

Can you confirm if it is going through one of the above?

What codec are you using?

Do you have any network acl's / Firewall issues that may be blocking any traffic?

Hope that helps

Scott


On Wed, Mar 27, 2013 at 10:31 PM, Deepak Maggo <dmaggo at ipcelerate.com<mailto:dmaggo at ipcelerate.com>> wrote:
Hi Scott,

1. Yes I am absolutely sure I verified it through Wireshark. packets are coming from Call Manager IP when I make call from IP phone (with BIB enabled) to PSTN through MGCP gateway (in this case I am receiving all RTP packets from both Agent & Customer voice stream).
2. By alternate packets I mean that when I make a call from IP Phone to IP Phone then under Wireshark on recording server it shows that RTP packets coming from the IP phone with BIB enabled but the sequence number of RTP packets is not continues, RTP packet with alternate  sequence number is missing because of this my recording server giving me choppy recordings. As you can see below RTP packet with Seq=20857 is missing and it is consistent through out the stream of RTP packets from phone  (.163 is IP Phone and .201 is Recording server).

No.     Time           Source                Destination           Protocol Length Info
     84 17.296525000   10.X.X.163          10.X.X.207          RTP      214    PT=ITU-T G.711 PCMU, SSRC=0x1603A8EB, Seq=20856, Time=209014710

No. Time Source Destination Protocol Length Info
98 17.336996000 10.X.X.163          10.X.X.207          RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x1603A8EB, Seq=20858, Time=209015030

I tried with CUCM 9.0 as well but I am getting the same response.


Thanks & Regards,
Deepak Maggo


On Wed, Mar 27, 2013 at 11:29 PM, Scott Voll <svoll.voip at gmail.com<mailto:svoll.voip at gmail.com>> wrote:
1.  are you sure it's not coming from the GW?
2.  What are alternate RTP packets.

What software package are you using for the recording?

BIB in CM 7.x notes:

BIB on device needs to be enabled
on the line you need a set your recording option to auto, setup your recording profile and CSS
you will need to associate the device being used to the Application user for the app your using.
You can not have shared lines in my experience (may be application specific)

BIB in CM 8 is much better.

Setup EM users then you can just assign the EM profile to the application user and don't have to worry about which phones are associated.

Hope that helps

Scott



On Wed, Mar 27, 2013 at 10:34 AM, Deepak Maggo <dmaggo at ipcelerate.com<mailto:dmaggo at ipcelerate.com>> wrote:
Hi,

I am facing the following problems when configuring my CUCM 7.0 to record calls using BIB feature:

1. When IP phone makes call to PSTN (using MGCP Gateway) then I receive the packets but they are coming from CUCM instead of from IP phone.

2. When IP phone makes call to IP Phone (on same CUCM) then I do receive the packets from IP phone but its sending only alternate RTP packets.

I'll appreciate any help to solve this issue.

Thanks in advance.


Thanks & Regards,
Deepak Maggo


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