7. RTP Relay
Brekeke SIP Server decides whether to do RTP relay or not automatically for each session. The Administrator can also specify whether Brekeke SIP Server should do RTP relay or not using a Dial Plan. RTP relay will be enabled automatically for the session where NAT traversal is handled.
The following handling variables are related to RTP relay:
$rtp = true or false
Whether to apply RTP Relay or not
$ifdst = IP address or FQDN
The interface address of Brekeke SIP Server for communicating with the session’s destination (the callee).
This address is used for receiving RTP packets from the destination UA.
$ifsrc = IP address or FQDN
The interface address of Brekeke SIP Server for communicating with the session’s source (the caller).
This address is used for receiving RTP packets from the source UA.
Ex 1. Enabling RTP Relay if the callee’s name prefix is “9”
| Matching Patterns | Deploy Patterns |
$request = ^INVITE To = sip:9.+@ |
$rtp = true |
Ex 2. Enabling RTP Relay and using PCMA as the codec if the callee’s SIP URI is sip:user@host
| Matching Patterns | Deploy Patterns |
$request = ^INVITE $geturi(To) = sip:user@host |
$rtp = true &net.rtp.audio.payloadtype = 8 |
Ex 3. Enabling RTP Relay and assigning the range of ports from 10000 to 10100 if the is call from 192.168.0.1
| Matching Patterns | Deploy Patterns |
$request = ^INVITE $addr = 192\.168\.0\.1$ |
$rtp = true &net.rtp.port.min = 10000 &net.rtp.port.max = 10100 |