1. Setting Info
1.1 Account Info
- Include: SIP Username – Password – Server (Domain)
- Usage: For provisioning (Register, Make Calls)
Options for Caller ID:
- Specific numbers: purchased numbers, rebranding numbers
- Anonymous: Private
- SIP number: the system will follow basic PBX caller ID settings
- PBX setting: the system will follow advanced PBX caller ID settings
When client sends incorrect caller ID, this number will be used.
1.2 Basic SIP Configuration
This part gives you the basic SIP information:
- Supported Codes: G.729, GSM, G.711A, G.711U
- Dial Pattern: indicates how to dial a number/make a call from SIP
- <Country Code> + <Area Code> + <Phone Numbers>
- SIP Protocol: Transport protocol: TCP vs UDP
- TCP gives more control
- UDP lighter
- TLS (secure)
- SIP Registration Period: 3600s
- Audio Format: RTP
- RTP Payload Size: 20ms
- DTMF: only supports RFC2833. (IVR)
1.3 Firewall Configuration
A firewall is a network security system that monitors and controls the incoming and outgoing network traffic based on predetermined security rules.
- Voice call = Audio + Signaling
*Make sure Firewall allows traffic following the rules mentioned in the SIP App:
Signaling (SIP):
- IP Addresses: 54.251.119.119
- Ports: 5060 (UDP/TCP)
- Direction: Both incoming and outgoing
Audio (RTP):
- IP Addresses: 54.251.255.196 to 54.251.255.211
- Ports: 10000-30000 (UDP)
- Direction: Both incoming and outgoing
1.4 NAT Configuration (Optional. Use only if PBX is behind NAT)
- NAT Traversal: STUN
- STUN Server: stun.hoiio.com
- STUN Port: 3478-3479
- STUN Protocol: UDP/TCP
- Direction: Both incoming and outgoing
1.5 None-Supported Features
- Call Transfer
- SIP Refer
- SIP Subscribe
- SIP Message
- SIP Publish
- Session Timers
- P-Asserted-Identity
- Remote-Party-ID
- Voice Activity Detection
- SIP ALG
2. Subscription Info
