I am using 10.0 version of CUCM. I am facing this issue. I have an user who uses both cisco jabber and deskphone. The user is configured as call center agent.
a. When I make a call and she answers the call through deskphone---> she can hear me and no issues.
b. When I make a call and she answers the call through Jabber by clicking the pop up window from jabber---> The call is getting answered but no voice on either sides.
On jabber I checked the USE MY COMPUTER FOR Calls and its selected. What could be the issue?
We also use Cisco's IP Communicator - and it works flawlessly. With Jabber 2-way audio works well when you dial internally and when receiving calls internally and externally. Only time the issue presents itself is when dialing to an external number. the call is picked-up but the audio does not work both ways.
thank you for such a quick response!
I have a couple ideas. It could be either a codec issue (Jabber likes G.722), or a firewall issue. The first thing I would try would be to do a wireshark capture on a VPN PC and get all the info regarding what happened. It may leap out at you in the SIP messages. Failing that, you can try to chase it down using info from the voice gateway (I am going to assume a Cisco CUBE router) and the firewall. The end points showing connection means the CUCM has told them to establish an RTP stream. It doesn't mean the RTP is actually getting there. I would troubleshoot this by bringing up an outside call and leaving it up. While the call is up, determine the call ID like this:
sh call act vo br | i pid 37AD : 1172619 2226758050ms.1 (14:59:21.506 EDT Thu Jul 22 2021) +1010 pid:78221 Answer 4075551212 active 37AD : 1172621 2226759050ms.1 (14:59:22.502 EDT Thu Jul 22 2021) +10 pid:78222 Originate 4075551313 active
That is an excerpt of the command with the numbers obfuscated. Now I want to see the IP's involved in the conversion. I'll use the call ID to do that:
CUBE-4K#sh call act vo id 37AD | i IP SIP call-legs: 2 VOIP: RemoteIPAddress=22.214.171.124 RemoteSignallingIPAddress=126.96.36.199 RemoteMediaIPAddress=188.8.131.52 VOIP: RemoteIPAddress=192.168.0.1 RemoteSignallingIPAddress=192.168.0.1 RemoteMediaIPAddress=192.168.0.2 SIP call-legs: 2
IP's obfuscated, but you see that the provider is using the same IP for both signalling and media. On the inside 192.168.0.1 is the CUCM (signalling) and 192.168.0.2 is the IP of the Jabber client. Now I have IP information I can use to look at the firewall and see if anything got blocked. I can also check and make sure that a media stream got started and what codec it is using.
CUBE-4K#sh call act vo id 37AD | i Code CoderTypeRate=g711ulaw CodecBytes=160 CoderTypeRate=g711ulaw CodecBytes=160
You could skip the filters ("i") and just grab all the info about the call to examine later. That should give you a starting point. I would also experiment with taking one Jabber client and force MTP on. You will need to make that there is an MTP in the MRGL that this client is using.
Hello Elliot Dierksen
Do you have a step-by-step process that we can follow to troubleshoot this type of issue and steps to modify our firewall?
That is what I was attempting to do with my previous post. Whatever firewall you are using must have a log. Bring up a call and leave it even if the audio isn't going through. Look at the firewall log for any deny messages.