Number of simultaneous Asynchronous Operations
IP Address and optional port to bind to for this transport
File containing a list of certificates to read (TLS ONLY)
Path to directory containing a list of certificates to read (TLS ONLY)
Certificate file for endpoint (TLS ONLY)
Preferred cryptography cipher names (TLS ONLY)
Domain the transport comes from
External IP address to use in RTP handling
External address for SIP signalling
External port for SIP signalling
Method of SSL transport (TLS ONLY)
Network to consider local (used for NAT purposes).
Password required for transport
Private key file (TLS ONLY)
Protocol to use for SIP traffic
Require client certificate (TLS ONLY)
Must be of type 'transport'.
Require verification of client certificate (TLS ONLY)
Require verification of server certificate (TLS ONLY)
Enable TOS for the signalling sent over this transport
Enable COS for the signalling sent over this transport
The timeout (in milliseconds) to set on WebSocket connections.
Allow this transport to be reloaded.
Configuration Option Descriptions
If a websocket connection accepts input slowly, the timeout for writes to it can be increased to keep it from being disconnected. Value is in milliseconds; default is 100 ms.
Allow this transport to be reloaded when res_pjsip is reloaded. This option defaults to "no" because reloading a transport may disrupt in-progress calls.
A way of creating an aliased name to a SIP URI
Value used in Max-Forwards header for SIP requests.
The interval (in seconds) to send keepalives to active connection-oriented transports.
The maximum amount of time from startup that qualifies should be attempted on all contacts. If greater than the qualify_frequency for an aor, qualify_frequency will be used instead.
Must be of type 'global'.
Value used in User-Agent header for SIP requests and Server header for SIP responses.
When set, Asterisk will dynamically create and destroy a NoOp priority 1 extension for a given peer who registers or unregisters with us.
Endpoint to use when sending an outbound request to a URI without a specified endpoint.
Enable/Disable SIP debug logging. Valid options include yes
no or a host address
The order by which endpoint identifiers are processed and checked. Identifier names are usually derived from and can be found in the endpoint identifier module itself (res_pjsip_endpoint_identifier_*)
When Asterisk generates an outgoing SIP request, the From header username will be set to this value if there is no better option (such as CallerID) to be used.
This documentation was imported from Asterisk Version GIT-13-0985f44d35c494