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Diffstat (limited to 'asterisk-conf/rtp.conf')
| -rw-r--r-- | asterisk-conf/rtp.conf | 162 |
1 files changed, 162 insertions, 0 deletions
diff --git a/asterisk-conf/rtp.conf b/asterisk-conf/rtp.conf new file mode 100644 index 0000000..19f3486 --- /dev/null +++ b/asterisk-conf/rtp.conf @@ -0,0 +1,162 @@ +; +; RTP Configuration +; +[general] +; +; RTP start and RTP end configure start and end addresses +; +; Defaults are rtpstart=5000 and rtpend=31000 +; +rtpstart=10000 +rtpend=10100 +; +; 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 recognized source of the RTP stream. Strict RTP qualifies RTP +; packet stream sources before accepting them upon initial connection and +; when the connection is renegotiated (e.g., transfers and direct media). +; Initial connection and renegotiation starts a learning mode to qualify +; stream source addresses. Once Asterisk has recognized a stream it will +; allow other streams to qualify and replace the current stream for 5 +; seconds after starting learning mode. Once learning mode completes the +; current stream is locked in and cannot change until the next +; renegotiation. +; Valid options are "no" to disable strictrtp, "yes" to enable strictrtp, +; and "seqno", which does the same thing as strictrtp=yes, but only checks +; to make sure the sequence number is correct rather than checking the time +; interval as well. +; 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 +; +; Enable sRTP replay protection. Buggy SIP user agents (UAs) reset the +; sequence number (RTP-SEQ) on a re-INVITE, for example, with Session Timers +; or on Call Hold/Resume, but keep the synchronization source (RTP-SSRC). If +; the new RTP-SEQ is higher than the previous one, the call continues if the +; roll-over counter (sRTP-ROC) is zero (the call lasted less than 22 minutes). +; In all other cases, the call faces one-way audio or even no audio at all. +; "replay check failed (index too old)" gets printed continuously. This is a +; software bug. You have to report this to the creator of that UA. Until it is +; fixed, you could disable sRTP replay protection (see RFC 3711 section 3.3.2). +; This option is enabled by default. +; srtpreplayprotection=yes +; +; Whether to enable or disable ICE support. This option is enabled by default. +; icesupport=false +; +; Hostname or address for the STUN server used when determining the external +; IP address and port an RTP session can be reached at. The port number is +; optional. If omitted the default value of 3478 will be used. This option is +; disabled by default. Name resolution will occur at load time, and if DNS is +; used, name resolution will occur repeatedly after the TTL expires. +; +; e.g. stundaddr=mystun.server.com:3478 +; +; stunaddr= +; +; Some multihomed servers have IP interfaces that cannot reach the STUN +; server specified by stunaddr. Blacklist those interface subnets from +; trying to send a STUN packet to find the external IP address. +; Attempting to send the STUN packet needlessly delays processing incoming +; and outgoing SIP INVITEs because we will wait for a response that can +; never come until we give up on the response. +; * Multiple subnets may be listed. +; * Blacklisting applies to IPv4 only. STUN isn't needed for IPv6. +; * Blacklisting applies when binding RTP to specific IP addresses and not +; the wildcard 0.0.0.0 address. e.g., A PJSIP endpoint binding RTP to a +; specific address using the bind_rtp_to_media_address and media_address +; options. Or the PJSIP endpoint specifies an explicit transport that binds +; to a specific IP address. Blacklisting is done via ACL infrastructure +; so it's possible to whitelist as well. +; +; stun_acl = named_acl +; stun_deny = 0.0.0.0/0 +; stun_permit = 1.2.3.4/32 +; +; For historic reasons stun_blacklist is an alias for stun_deny. +; +; Whether to report the PJSIP version in a SOFTWARE attribute for all +; outgoing STUN packets. This option is enabled by default. +; +; stun_software_attribute=yes +; +; Hostname or address for the TURN server to be used as a relay. The port +; number is optional. If omitted the default value of 3478 will be used. +; This option is disabled by default. +; +; e.g. turnaddr=myturn.server.com:34780 +; +; turnaddr= +; +; Username used to authenticate with TURN relay server. +; turnusername= +; +; Password used to authenticate with TURN relay server. +; turnpassword= +; +; An ACL can be used to determine which discovered addresses to include for +; ICE, srflx and relay discovery. This is useful to optimize the ICE process +; where a system has multiple host address ranges and/or physical interfaces +; and certain of them are not expected to be used for RTP. For example, VPNs +; and local interconnections may not be suitable or necessary for ICE. Multiple +; subnets may be listed. If left unconfigured, all discovered host addresses +; are used. +; +; ice_acl = named_acl +; ice_deny = 0.0.0.0/0 +; ice_permit = 1.2.3.4/32 +; +; For historic reasons ice_blacklist is an alias for ice_deny. +; +; The MTU to use for DTLS packet fragmentation. This option is set to 1200 +; by default. The minimum MTU is 256. +; dtls_mtu = 1200 +; +[ice_host_candidates] +; +; When Asterisk is behind a static one-to-one NAT and ICE is in use, ICE will +; expose the server's internal IP address as one of the host candidates. +; Although using STUN (see the 'stunaddr' configuration option) will provide a +; publicly accessible IP, the internal IP will still be sent to the remote +; peer. To help hide the topology of your internal network, you can override +; the host candidates that Asterisk will send to the remote peer. +; +; IMPORTANT: Only use this functionality when your Asterisk server is behind a +; one-to-one NAT and you know what you're doing. If you do define anything +; here, you almost certainly will NOT want to specify 'stunaddr' or 'turnaddr' +; above. +; +; The format for these overrides is: +; +; <local address> => <advertised address>,[include_local_address] +; +; The following will replace 192.168.1.10 with 1.2.3.4 during ICE +; negotiation: +; +;192.168.1.10 => 1.2.3.4 +; +; The following will include BOTH 192.168.1.10 and 1.2.3.4 during ICE +; negotiation instead of replacing 192.168.1.10. This can make it easier +; to serve both local and remote clients. +; +;192.168.1.10 => 1.2.3.4,include_local_address +; +; You can define an override for more than 1 interface if you have a multihomed +; server. Any local interface that is not matched will be passed through +; unaltered. Both IPv4 and IPv6 addresses are supported. |