08-21-2009 03:39 AM - edited 03-15-2019 07:27 PM
Hello,
I have AS5350 and Asterisk IP PBX connected to each other. How to set the RTP ports range using for the SIP media flows at the cisco side ?
08-21-2009 04:14 AM
SIP and RTP are two different sets of protocol.
SIP is an industry standard and uses 5060/61 (TCP/UDP) ports.
RTP has a broad range of ports assigned 16384 - 32767 UDP. However different vendors use different ports (e.g. CUCM uses only a number 24576-32767/UDP) hence you may want to check the ASterisk Documentation to make sure you open only concerned ports.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/6_1/61plrev1.pdf
Hope this helps.
Please rate all helpful posts.
Regards
Wilson Samuel
08-21-2009 04:54 AM
Hi,
Cisco GWs use the full 16384 - 32767 UDP range.
If you need more specific firewalling you'll need a protocol-aware FW that will open up udp pin-holes based on what was negotiated during the call-setup session.
If you mean to limit the number of ports used to source RTP from your AS - I don't have any idea if it's possible...
BR,
08-21-2009 06:54 AM
There are no hard-standards that you can guarantee for this.
By default, the gateway will use TCP/UDP 5060, and for SIP-TLS TCP 5061.
The gateway will advertise ports between 16384-32768.
This is no means guarantees that the SIP provider will also. They frequently will use ports from anywhere in the 4000-40000 range. I would probe Asterisk about their UDP port range.
-nick
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