sipXcom:RTP packets not sending

[Linu Kurian]

Hi Team,
I am using "sipxcom-18.12-4981.98933-x86_64". I am using virtual phones to perform the voip calls. I created the users in sipXcom and the phones are registered in sipXcom servers. When I start calling I am getting an error "No Media Present". I captured the packets during the call and found that SIP packets are sending and recieving but no RTP packets(Actual media) is not sending. Can we do anything with the config?
Please help
Thanks,
Linu

[Matt Keys]

Check that system - services - nat traversal is set to 'specify IP' rather than using stun and specify the private IP of the server. Next check beneath system - settings - internet calling that networks local to the server are listed and remove any extras. Reboot and retest.

[Linu Kurian]

Hi ,
Can you please specify, which server ip i need to mention under  system - services - nat traversal

[Matt Keys]

Typically it is the private (local lan) IP of the server. 

[Linu Kurian]

Hi,
I have added the sipXcom server ip into the  system - services - nat. But rtp not passing. The entire system is deployed in separate separate private networks and connected using a firewall, not connected to Internet.
Below are the registration entries from sipXcom server,

"NETSCOUT nGeniusPULSE"<sip:10...@probe.net> "NETSCOUT nGeniusPULSE"<sip:10...@10.0.0.66:5060;ob;x-sipX-nonat> 603

"NETSCOUT nGeniusPULSE"<sip:10...@probe.net> "NETSCOUT nGeniusPULSE"<sip:10...@10.0.0.15:5060;ob;x-sipX-nonat> 613

"NETSCOUT nGeniusPULSE"<sip:10...@probe.net> "NETSCOUT nGeniusPULSE"<sip:10...@10.0.0.38:38007;ob;x-sipX-nonat> 531

"NETSCOUT nGeniusPULSE"<sip:10...@probe.net> "NETSCOUT nGeniusPULSE"<sip:10...@10.0.0.62:5060;ob;x-sipX-nonat> 666

Below is the settings,

NAT Traversal  sipXcom.probe.net

Hide Advanced Settings

NAT

Address type

Specify IP address

Public IP address

192.168.7.5

When the server is deployed behind a NAT, the "Public IP address" field must be set to the Internet-facing IP address of the NAT / firewall device fronting the server.

SIP Port

5060

TLS SIP Port

5061

Start RTP port

3000

First port of the UDP port range allocated to the sipXrelay process for the purposes of relaying media traffic for NAT traversal.

End RTP port

6000

Last port of the UDP port range allocated to the sipXrelay process for the purposes of relaying media traffic for NAT traversal.

Please advise, If I need to change anything 

Thanks,
Linu

  

[Matt Keys]

I would guess it's a network or firewall issue. You'll need to pcap both ends of a test call to see where the RTP is stopping. 

[Linu Kurian]

Also, In our network there is no direct connectivity from caller to Callee, but can connect each other through sipXcom server. Will this create any issue?.
 I have captured the packets on both caller/callee agents, sipXcom server and the router there is no RTP packets found.

Thanks,
Linu