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 |
Video Support
Obelisk includes support for SIP video calls using the H.264 codec by default.
Most Cloud/SaaS SIP providers do not support video calls, so we strongly recommend verifying video capabilities with your SIP provider.
Cisco Call Manager
Compatibility with Cisco Unified Communications Manager (CUCM) has been tested and verified. To enable video calls, Early Offer must be enabled in the SIP Profile on CUCM.
Cisco RoomKit
Cisco RoomKit devices have been tested and verified to work with Obelisk, both via Asterisk and CUCM.
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 | 6000000 | Maximum allowed video bitrate to the caller in bits/s |
encode_video_at_half_bitrate |
bool |
no | false |
Encode video at half the bitrate advertised by the caller. This ensures compatibility with devices that incorrectly report their supported H.264 maximum bitrate for both sending and receiving, instead of just receiving. |
If ÃŽd 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