Hi,
I'm all new regarding the Prognosis VoIP Monitor.
So far i got it installed, and i can see that Prognosis is getting data concerning stations, trunks, alarms etc. But i have no voicestreams (RTCP?)....

Setup is as follows.
CM located at location 1 network 10.1.2.x
CM LSP located at location 2 network 10.1.16.x
Prognosis located at location 3, network 10.1.15.x with NAT remapped to 172.25.61.x
All networks are connected together via an Operator MPLS network
CM is setup to send RTCP to 10.1.15.x wihich is then translated with NAT as described above.

The setup is like this as we are a businesspartner doing all the monitoring for the customer, who hosts the telephony platform themselves and we host all manament and monitoring platforms.
Im wondering if its network related? or if i just manage something wrong?

The RTCP comes directly from the phones, not the CM. So the PROGNOSIS server must be able to be contacted by every phone.

This could be a NAT issue that the RTCP from the phones is not getting to the PROGNOSIS server. You can verify if PROGNOSIS is receiving RTCP, by clicking from the main PBX screen (at the top of the display) Configuration the RTCP. You will see a summary of the RTCP packets being received.

Hi Paul,

I'm having a similar issue. But I do NOT have NAT set up for the Prognosis. The Prognosis sits on the same VLAN as Avaya CM. I'm able to see hits on the Avaya RTCP configuration however, I couldn't see anything when I click on the Finished Call (by extension). I'm expecting similar result as per the VQ demo shown.

Setup as follow,
CM located at location A network 172.17.77.x
Prognosis located at location A, network 172.17.77.xxx
CM LSP located at location B network 172.19.22.x
IP Phones located at both location A & B which has reachability to to the Prognosis at Location A (we are using softphone and ping from my laptop to Prognosis suceed)

All networks are connected together via an Operator MPLS network
CM is setup to send RTCP to 172.17.77.xxx.

Regards,
Nicholas

One way to view if you've cofigured the RTCP destinations correctly is to view the RTCP display from within Voice Monitor. The drill down to view this is located in the Configuration display.

So, start from the main display for your Avaya PBX and drill down into the configuration display (the drill down is in the upper right hand corner of the main display). From there click on the RTCP drill down. It will show you where the switch is sending the RTCP packets. This should be the IR Voice Monitor server.

Yeah, the RTCP seems to be working fine. However, I started wondering if the voip monitoring software only work on device type 9640? Finished calls (voice quality trace) for device type 9640 works fine but not 9630 and 9650.

Does anyone has any idea about this? (Refer to my previous post). Thanks.

It will work on the 9640's unless you are running the SIP load.

Yes it works on the 9640's but NOT the 9630 and 9650. Can you confirm if this is the behavior it supposed to be?

I am not aware of any issues with the 9630 or 9650. I will be investigating but am fairly sure that all should work fine.

Alright..will be waiting the result of your finding....

Can you tell me what firmware you are running on the phones and what version of CM you are running? Also, do you have any MedPro's with anything less than v55?

Nicholas,

There are no problems with 9620 and 9630 phones and see no issue why 9650's could be an issue as they use the same H.323/RTP stacks.

When you say you are receiving RTCP information, can you please clarify a few things? Things to investigate:

1. Are there any or missing voice streams on the main PBX display page? If so, verify the RTCP configuration. Go to the AV-PBX screen, click on Configuration then the RTCP link. This will show the number of RTCP packets received and also the configuration per ip-network-region of the RTCP destination address. You should check the correct address for each network region. If some are showing, but not all of them are showing, it is probably a issue with a particular network segment so you should check the setting for each network region.
2. If the configuration is correct, use the RTCP Configuration screen to check the count of received RTCP packets, this will determine if Prognosis is receiving RTCP packets.
3. If Prognosis is not receiving any RTCP packets, check that the server is receiving them. Install Wireshark and perform a packet capture filtering on port 5005 to see if RTCP packets are being received - they may initially show up as UDP port 5005, but you can 'decode as' RTCP in Wireshark.
4. If the Prognosis server is not receiving any RTCP packets, there is potentially a firewall issue, on the PC or in the network, or a routing/VLAN issue. Please check connectivity to the phone from the Prognosis server. Firstly try and ping/traceroute the phone from the Prognosis server. If this is successful, you may need to investigate any firewall rules for UDP.

Can you please inform us of which step you are encountering issues?

Thanks,
Paul

I will keep you updated by Monday. Thanks

Hi Paul,

To answer your questions.

1. Yes, I realised there were missing voice streams on the main PBX display page for softphones. RTCP configuration, Ip network region of the RTCP destinaation are correct.

2. Yes, the count was incorrect as I mentioned the RTCP only capture voice stream for calls made uses hard phone.

3. I have done the wireshark capturing for softphone, media gateway, hard phone and prognosis. At 1st, the wireshark did not capture any traffic for port 5005 for calls made by softphone. After upgrading my one-X communicator to release 5.2 with SP3 (5.2.0.19), I was able to see traffic uses port 5005 sending from the softphone to the Prognosis server. However, I'm still unable to see any active voice stream or finished calls QoS result on Prognosis server.

4. I have also done the wireshark capturing on the Prognosis server and it shows the server is communicating with the softphone on port 5005. This means, the network layer is pretty much clear and both ends are communication on port 5005.

Regards,
Nicholas

Further to my findings, I found out the differences between the softphone & hardphone RTCP packet

9640(Hardphone)
SDES items
Type: CNAME (user and domain) (1)
Length:27
Text: ext57002@172.19.18.197:2074
Type: PHONE (phone number) (4)
Length: 5
Text: 57002
Type: TOOL (name/version of source app) (6)
Length:37
Text: Avaya IP Telephone (ha96xxu3_00.bin)
Type: END (0)

9630 (Softphone)
SDES items
Type: CNAME (user and domain) (1)
Length:27
Text: sip:<unknown>@172.19.18.111
Type: PHONE (phone number) (4)
Length: 8
Text: 10714:30
Type: TOOL (name/version of source app) (6)
Length:29
Text: Avaya PC Audio Component R6.0T
Type: END (0)

And I found out some information from Avaya documentation:

RSVP & RTCP Monitor Server - 9600 Series SIP IP Telephones do not use RSVP (Resource
ReSerVation Protocol) software to provide real-time monitoring and historical data of audio
quality for VoIP calls. 9600 Series SIP IP Telephones do support an RTCP Monitor Server.

QoS - Unlike H.323 telephones, 9600 Series SIP IP Telephones do not use Avaya
Communication Manager to set Quality of Service (QoS). The SIP IP telephones use the
parameters L2QAUD, L2QSIG, DSCPAUD, and DSCPSIG

The strange thing is my 9630 softphone details is showing Signaling Protocol:H.323 instead of SIP (above RTCP info is showing SIP instead of H.323)
Any idea?


Regards,
Nicholas

Nicholas,

We have raised this as an issue with Avaya a few months ago and are working with them to resolve this on the One-X Communicator.

I will get back to you as soon as I know an ETA on when the fix will be available.

Paul

Paul,

Thanks for the info. I'm in the midst of getting the evaluation IPTM and IPTMR copy and will be installed in our environment for testing. If this issue has not been solved by Avaya, I'm afraid during our IPTM and IPTMR test later, it will hit the same issue.

Rgds,
Nicholas

BTW, this is also happen to one-X Agent...