asterisk/configs/rtp.conf.sample

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;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000 ; Milliseconds between rtcp reports
;(min 500, max 60000, default 5000)
;
; Enable strict RTP protection. This will drop RTP packets that
; do not come from the source of the RTP stream. This option is
; enabled by default.
; strictrtp=yes
;
; Number of packets containing consecutive sequence values needed
; to change the RTP source socket address. This option only comes
; into play while using strictrtp=yes. Consider changing this value
; if rtp packets are dropped from one or both ends after a call is
; connected. This option is set to 4 by default.
; probation=8
;
; Whether to enable or disable ICE support. This option is enabled by default.
; icesupport=false
;
; Address to use for the STUN server when determining the external IP address and port
; an RTP session can be reached at. This option is disabled by default.
; stunaddr=
;
; Address to use for the TURN relay server when creating a TURN relay session. This option
; is disabled by default.
; turnaddr=
;
; Port used to contact the TURN relay server on. This option is set to 34780 by default.
; turnport=34780
;
; Username used to authenticate with TURN relay server.
; turnusername=
;
; Password used to authenticate with TURN relay server.
; turnpassword=