SIP SIPS URI Support
From Yate Documentation
Notes:
- There is no automatic handling for incoming calls using SIPS URI. They must be explicitly forwarded as SIPS URI
- When a SIPS URI is used to send a request transport defaults to TLS unless explicitly set to another value (UDP/TCP).
Using with lines (accounts)
A line can be configured to register using a SIPS URI by adding a sips=true parameter. accfile.example:
[my_sips_line] ; other params ... sips=true
A line can be configured to register using a SIPS URI on non TLS transport.
This may be used when transport to registrar is already secured (e.g. with IPsec).
[my_sips_line] ; other params ... ip_transport=udp sips=true
Routing
Incoming SIPS call sent using SIPS:
${sip_uri}^sips:=sip/sips:${called}@example.com
Outgoing call sent on a line using SIPS URI. The module will generate a SIPS URI:
^123=line/${called};line=my_sips_line
Outgoing call sent on a line using SIPS URI but target set to a SIP/TEL URI. The module won't generate a SIPS URI:
^123=sip/sip:${called}@example.com;line=my_sips_line ^123=sip/tel:+${called};line=my_sips_line
Outgoing call using SIPS URI on non TLS transport:
^123=sip/sips:${called}@example.com;oip_transport=udp
A line's SIPS URI usage may be overridden from routing by a sips parameter.
Outgoing call send using some line. Disable SIPS URI usage even if configured for used line:
^123=line/${called};line=my_line;sips=no