SIP
Supported codecs
The Obelisk supports the following Codecs:
| Codec | Clock-Rate |
|---|---|
| G.711 (PCMA) | 8,000 |
| G.711 (PCMU) | 8,000 |
| G.722 | 8,000 |
| H.264 | 90,000 |
Configuration
The section in the configuration file is called sip.
| Field | Type | Required | Default value | Description |
|---|---|---|---|---|
addr | string | yes | - | The local IP address to bind to (0.0.0.0 binds to every address) |
port | int | yes | - | The port to bind to (usually 5060) |
id | string | no | See below | The ID of this SIP endpoint in the format sip:<username>@<host>, used for creating the From/To header in SIP REGISTER messages |
contact | string | no | See below | The Contact of this SIP endpoint in the format sip:<username>@<host>, used for creating the SIP Contact header |
username | string | no | none | The username to register with the SIP provider |
password | string | no | none | The password to register with the SIP provider |
realm | string | no | none | The realm of the given username/password pair |
registrar | string | no | none | The SIP URI of the registrar in the format sip:<domain> |
outbound_proxy | string | no | none | The SIP proxy to send all requests to in the format sip:<domain> |
nat_ping_delta | string | no | 30 seconds | Seconds between ping and pong to keep the NAT binding alive |
stun_server | string | no | none | The host and optional port of the STUN server in the format <host>[:<port>] |
enforce_qop | bool | no | false | true to enforce quality of protection on SIP authentication |
offer_srtp | bool | no | false | true to offer SRTP as media transport when obelisk is creating the initial SDP offer |
rtp_port_range | RTP port range | no | 40000-49999 | The port range for the SIP RTP/RTCP connections |
max_video_bitrate | int | no | 6000 | Maximum allowed video bitrate to the caller in kbits/s |
If ìd is not set, it is generated as sip:<username>@<registrar-host>. If no registrar is configured it is generated just like the contact field.
If contact is not set, it is generated as sip:<username>@<addr> where <addr> may be replaced by the public address discovered using the STUN server.
RTP port range
| Field | Type | Required | Default value | Description |
|---|---|---|---|---|
start | string | yes | - | The lower bound of the port range for the SIP RTP/RTCP connections |
end | string | yes | - | The upper bound of the port range for the SIP RTP/RTCP connections |
Example
[sip]
addr = "0.0.0.0"
port = 5060
id = "sip:alice@example.org"
contact = "sip:alice@192.168.1.100"
username = "user"
password = "pass"
realm = "asterisk"
registrar = "sip:sip.example.org"
stun_server = "stun.example.org:3478"
[sip.rtp_port_range]
start = 40000
end = 49999