Mise à jour après premières vagues de gros tests
[asterisk-config-auf.git] / etc-asterisk / rtp.conf
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1;
2; RTP Configuration
3;
4[general]
5;
6; RTP start and RTP end configure start and end addresses
7;
8; Defaults are rtpstart=5000 and rtpend=31000
9;
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10; juste après /proc/sys/net/ipv4/ip_local_port_range
11rtpstart=61001
12rtpend=62000
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13;
14; Whether to enable or disable UDP checksums on RTP traffic
15;
16;rtpchecksums=no
17;
18; The amount of time a DTMF digit with no 'end' marker should be
19; allowed to continue (in 'samples', 1/8000 of a second)
20;
21;dtmftimeout=3000
22; rtcpinterval = 5000 ; Milliseconds between rtcp reports
23 ;(min 500, max 60000, default 5000)
24;
25; Enable strict RTP protection. This will drop RTP packets that
26; do not come from the source of the RTP stream. This option is
27; disabled by default.
28; strictrtp=yes
29;
30; Number of packets containing consecutive sequence values needed
31; to change the RTP source socket address. This option only comes
32; into play while using strictrtp=yes. Consider changing this value
33; if rtp packets are dropped from one or both ends after a call is
34; connected. This option is set to 4 by default.
35; probation=8