163 lines
6.7 KiB
Plaintext
163 lines
6.7 KiB
Plaintext
;
|
|
; 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 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.
|