sipXcom:RTP packets not sending

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Linu Kurian

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Jul 19, 2021, 12:13:10 AMJul 19
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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

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Jul 19, 2021, 7:24:11 AMJul 19
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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

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Jul 19, 2021, 9:21:30 AMJul 19
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Hi ,
Can you please specify, which server ip i need to mention under  system - services - nat traversal

Matt Keys

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Jul 19, 2021, 10:13:57 AMJul 19
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Typically it is the private (local lan) IP of the server. 

Linu Kurian

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Jul 19, 2021, 11:04:49 PMJul 19
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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

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Jul 20, 2021, 8:08:14 AMJul 20
to sipxcom-users
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

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Jul 20, 2021, 10:57:17 AMJul 20
to sipxcom-users
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

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