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b9e888418e
Correct typos of the following word families: password excludes undesirable checksums through screening interpreting database causes initiation member busydetect defined severely throughput recognized counter require indefinitely accounts ASTERISK-29714 Change-Id: Ie8f2a7b274a162dd627ee6a2165f5e8a3876527e
163 lines
6.7 KiB
Plaintext
163 lines
6.7 KiB
Plaintext
;
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; RTP Configuration
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;
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[general]
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;
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; RTP start and RTP end configure start and end addresses
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;
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; Defaults are rtpstart=5000 and rtpend=31000
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;
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rtpstart=10000
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rtpend=20000
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;
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; Whether to enable or disable UDP checksums on RTP traffic
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;
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;rtpchecksums=no
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;
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; The amount of time a DTMF digit with no 'end' marker should be
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; allowed to continue (in 'samples', 1/8000 of a second)
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;
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;dtmftimeout=3000
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; rtcpinterval = 5000 ; Milliseconds between rtcp reports
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;(min 500, max 60000, default 5000)
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;
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; Enable strict RTP protection. This will drop RTP packets that do not come
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; from the recognized source of the RTP stream. Strict RTP qualifies RTP
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; packet stream sources before accepting them upon initial connection and
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; when the connection is renegotiated (e.g., transfers and direct media).
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; Initial connection and renegotiation starts a learning mode to qualify
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; stream source addresses. Once Asterisk has recognized a stream it will
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; allow other streams to qualify and replace the current stream for 5
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; seconds after starting learning mode. Once learning mode completes the
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; current stream is locked in and cannot change until the next
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; renegotiation.
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; Valid options are "no" to disable strictrtp, "yes" to enable strictrtp,
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; and "seqno", which does the same thing as strictrtp=yes, but only checks
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; to make sure the sequence number is correct rather than checking the time
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; interval as well.
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; This option is enabled by default.
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; strictrtp=yes
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;
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; Number of packets containing consecutive sequence values needed
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; to change the RTP source socket address. This option only comes
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; into play while using strictrtp=yes. Consider changing this value
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; if rtp packets are dropped from one or both ends after a call is
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; connected. This option is set to 4 by default.
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; probation=8
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;
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; Enable sRTP replay protection. Buggy SIP user agents (UAs) reset the
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; sequence number (RTP-SEQ) on a re-INVITE, for example, with Session Timers
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; or on Call Hold/Resume, but keep the synchronization source (RTP-SSRC). If
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; the new RTP-SEQ is higher than the previous one, the call continues if the
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; roll-over counter (sRTP-ROC) is zero (the call lasted less than 22 minutes).
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; In all other cases, the call faces one-way audio or even no audio at all.
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; "replay check failed (index too old)" gets printed continuously. This is a
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; software bug. You have to report this to the creator of that UA. Until it is
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; fixed, you could disable sRTP replay protection (see RFC 3711 section 3.3.2).
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; This option is enabled by default.
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; srtpreplayprotection=yes
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;
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; Whether to enable or disable ICE support. This option is enabled by default.
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; icesupport=false
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;
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; Hostname or address for the STUN server used when determining the external
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; IP address and port an RTP session can be reached at. The port number is
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; optional. If omitted the default value of 3478 will be used. This option is
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; disabled by default. Name resolution will occur at load time, and if DNS is
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; used, name resolution will occur repeatedly after the TTL expires.
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;
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; e.g. stundaddr=mystun.server.com:3478
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;
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; stunaddr=
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;
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; Some multihomed servers have IP interfaces that cannot reach the STUN
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; server specified by stunaddr. Blacklist those interface subnets from
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; trying to send a STUN packet to find the external IP address.
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; Attempting to send the STUN packet needlessly delays processing incoming
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; and outgoing SIP INVITEs because we will wait for a response that can
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; never come until we give up on the response.
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; * Multiple subnets may be listed.
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; * Blacklisting applies to IPv4 only. STUN isn't needed for IPv6.
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; * Blacklisting applies when binding RTP to specific IP addresses and not
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; the wildcard 0.0.0.0 address. e.g., A PJSIP endpoint binding RTP to a
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; specific address using the bind_rtp_to_media_address and media_address
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; options. Or the PJSIP endpoint specifies an explicit transport that binds
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; to a specific IP address. Blacklisting is done via ACL infrastructure
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; so it's possible to whitelist as well.
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;
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; stun_acl = named_acl
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; stun_deny = 0.0.0.0/0
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; stun_permit = 1.2.3.4/32
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;
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; For historic reasons stun_blacklist is an alias for stun_deny.
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;
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; Whether to report the PJSIP version in a SOFTWARE attribute for all
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; outgoing STUN packets. This option is enabled by default.
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;
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; stun_software_attribute=yes
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;
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; Hostname or address for the TURN server to be used as a relay. The port
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; number is optional. If omitted the default value of 3478 will be used.
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; This option is disabled by default.
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;
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; e.g. turnaddr=myturn.server.com:34780
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;
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; turnaddr=
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;
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; Username used to authenticate with TURN relay server.
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; turnusername=
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;
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; Password used to authenticate with TURN relay server.
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; turnpassword=
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;
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; An ACL can be used to determine which discovered addresses to include for
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; ICE, srflx and relay discovery. This is useful to optimize the ICE process
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; where a system has multiple host address ranges and/or physical interfaces
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; and certain of them are not expected to be used for RTP. For example, VPNs
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; and local interconnections may not be suitable or necessary for ICE. Multiple
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; subnets may be listed. If left unconfigured, all discovered host addresses
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; are used.
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;
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; ice_acl = named_acl
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; ice_deny = 0.0.0.0/0
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; ice_permit = 1.2.3.4/32
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;
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; For historic reasons ice_blacklist is an alias for ice_deny.
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;
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; The MTU to use for DTLS packet fragmentation. This option is set to 1200
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; by default. The minimum MTU is 256.
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; dtls_mtu = 1200
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;
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[ice_host_candidates]
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;
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; When Asterisk is behind a static one-to-one NAT and ICE is in use, ICE will
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; expose the server's internal IP address as one of the host candidates.
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; Although using STUN (see the 'stunaddr' configuration option) will provide a
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; publicly accessible IP, the internal IP will still be sent to the remote
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; peer. To help hide the topology of your internal network, you can override
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; the host candidates that Asterisk will send to the remote peer.
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;
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; IMPORTANT: Only use this functionality when your Asterisk server is behind a
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; one-to-one NAT and you know what you're doing. If you do define anything
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; here, you almost certainly will NOT want to specify 'stunaddr' or 'turnaddr'
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; above.
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;
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; The format for these overrides is:
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;
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; <local address> => <advertised address>,[include_local_address]
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;
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; The following will replace 192.168.1.10 with 1.2.3.4 during ICE
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; negotiation:
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;
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;192.168.1.10 => 1.2.3.4
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;
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; The following will include BOTH 192.168.1.10 and 1.2.3.4 during ICE
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; negotiation instead of replacing 192.168.1.10. This can make it easier
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; to serve both local and remote clients.
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;
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;192.168.1.10 => 1.2.3.4,include_local_address
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;
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; You can define an override for more than 1 interface if you have a multihomed
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; server. Any local interface that is not matched will be passed through
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; unaltered. Both IPv4 and IPv6 addresses are supported.
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