Asterisk config rtp.conf

rtp.conf

Configuration of Asterisk Real Time Protocol, RTP, media channels. RTP is used for SIP communication.

Details

  • 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!
  • Question2:
    • 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...

Related issues:
  • 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


Example


;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=10000
rtpend=20000


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.

Related:

See also


rtp.conf

Configuration of Asterisk Real Time Protocol, RTP, media channels. RTP is used for SIP communication.

Details

  • 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!
  • Question2:
    • 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...

Related issues:
  • 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


Example


;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=10000
rtpend=20000


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.

Related:

See also


Created by: oej, Last modification: Tue 12 of May, 2009 (14:10 UTC) by JustRumours
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