SIP Profile
SIP Profile/Interface is an application layer interface logically residing over a network interface. The SIP profile defines the transport addresses (IP address and port) upon which the LibreSBC receives and sends SIP messages and RTP packets as well as related rtp media and signalling setting. You can define a SIP profile for each network to which the LibreSBC is connected. SIP profile support UDP, TCP and TLS transport. In addition to defining a SIP Profile's IP address might be defined by NetAlias API.
Consider any setting that you want to add, following is the configuration parameters:
| Parameter | Category | Description |
|---|---|---|
| name |
string required
|
The name of profile (unique) |
| desc | string |
Description |
| user_agent | string |
The value will be displayed as User-Agent in SIP header. Default is LibreSBC
|
| sdp_user | string |
The username of the o= and s= fields in SDP body |
| local_network_acl | string |
Set the local network that refer from predefined acl |
| enable_100rel | bool |
Reliability - PRACK message as defined in RFC3262 |
| ignore_183nosdp | bool |
Just ignore SIP 183 without SDP body |
| sip_options_respond_503_on_busy | bool |
response 503 when system is in heavy load |
| disable_transfer | bool |
true mean disable call transfer |
| manual_redirect | bool |
how call forward handled, true mean it be controlled under libresbc contraints, false mean it be work automatically |
| enable_3pcc | bool |
determines if third party call control is allowed or not |
| enable_compact_headers | bool |
disable as default, true to enable compact SIP headers |
| enable_timer | bool |
true to support for RFC 4028 SIP Session Timers |
| session_timeout | int |
call to expire after the specified seconds |
| minimum_session_expires | int |
Value of SIP header Min-SE |
| dtmf_type | enum |
Dual-tone multi-frequency (DTMF) signal type rfc2833 info none
|
| media_timeout | int |
The number of seconds of RTP inactivity before SBC considers the call disconnected, and hangs up (recommend to use session timers instead), default value is 0 - disables the timeout. |
| rtp_rewrite_timestamps | bool |
set true to regenerate and rewrite the timestamps in all the RTP streams going to an endpoint using this SIP Profile, necessary to fix audio issues when sending calls to some paranoid and not RFC-compliant gateways |
| context |
string required
|
The dialplan context of SIP profile |
| sip_port | int |
Port to bind to for SIP traffic |
| sip_address |
string required
|
IP address via NetAlias use for SIP Signalling |
| rtp_address |
string required
|
IP address via NetAlias use for RTP Media |
| tls | bool |
Enable TLS transport |
| tls_only | bool |
set True to disable listening on the unencrypted port for this connection |
| sips_port | int |
Port to bind to for TLS SIP traffic |