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Asterisk config rtp.conf
rtp.confConfiguration of Asterisk Real Time Protocol, RTP, media channels. RTP is used for SIP communication.
- on your router you might want to arrange both traffic shaping (QoS) and port forwarding (in case of NAT) for the RTP range that you chose
- for each RTP port, you also open RTCP port. Therefore a call can consume up to 4 RTP ports.
- the first port of the range should be even, so 10001 won't be used (use 10000 or 10002 instead); the last port must be uneven, and if you specify e.g. 10017 as last in range asterisk will actually use 10018, so be aware!
- maybe ports aren't released directly by Asterisk after the call has completed?
- does Asterisk allocate RTP ports for each member in a group dial (DIAL(SIP/device1&SIP/device2) before the actual call is established?
- check with "netstat -anup" or "netstat -anu" for open ports
- experience shows that often Asterisk seems to consume more RTP ports (or RTP port numbers) than one would expect, so it is most probably not a good idea to reduce the RTP port range to exactly 4 times the maximum number of concurrent calls...
- bug 14777 and bug 11257: Error "No RTP ports remaining. Can't setup media stream for this call."
- one possible cause/solution: check if ulimit is set high enough
- possible related: bug 8036
; RTP Configuration
; RTP start and RTP end configure start and end addresses
If you have a NAT or firewall between Asterisk and the server, you need to set these up to handle forwarding of the configured ports.
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