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AIVO Connect

SIP Network Information

Technical reference for AIVO Connect SIP trunking. Signaling addresses, transport protocols, media configuration, and firewall rules for integrating your PBX or SBC.

SIP Signaling Addresses

RegionFQDNPrimary IPSecondary IP
USsip.aivo.bz192.76.120.1064.16.250.10
Europesip-eu.aivo.bz185.246.41.140185.246.41.141
Australiasip-au.aivo.bz103.115.244.145103.115.244.146
Canadasip-ca.aivo.bz192.76.120.3164.16.250.13
SIP regions affect only the signaling path. Media anchoring is configured separately in your connection settings.
Outbound calls from the same IP or SIP username are limited to 50 calls per 3 seconds.

Inbound Calls

Register your SIP device on any region listed above. Inbound calls arrive from the regional SIP proxy. If you use an ACL or firewall, whitelist both primary and secondary IPs.

DNS Records

All SIP FQDNs support A, SRV, and NAPTR DNS record types.

Transport Protocols

ProtocolPort
UDP5060
TCP5060
TLS5061
WSS (WebRTC)443

Media (RTP)

RTP port range: 16384 to 65535 (UDP)

SRTP encryption is supported. Enable "Encrypted Media" on your SIP connection to require SRTP (RTP/SAVP) for all calls. WebRTC connections use DTLS-SRTP by default.

Firewall Whitelist Subnets

36.255.198.128/25
50.114.136.128/25
50.114.144.0/21
64.16.226.0/24
64.16.227.0/24
64.16.228.0/24
64.16.229.0/24
64.16.230.0/24
64.16.248.0/24
64.16.249.0/24
64.16.250.0/24
103.115.244.128/25
185.246.41.128/25
192.76.120.0/24

Codecs

  • G.722
  • G.711U (PCMU)
  • G.711A (PCMA)
  • G.729
  • Opus Supported for inbound and outbound. Inbound requires TLS or TCP transport.
  • T.38 Fax-over-IP. Enable T.38 on your SIP connection for fax pass-through.
  • H.264 Video codec. Available on video-enabled SIP connections.
  • telephone-event (RFC 2833) Used for DTMF tone relay.
We support ptime:20 for RTP.

Quick Start

  1. 1Log in to your AIVO Connect dashboard
  2. 2Create a SIP Connection (credential, IP, or FQDN auth)
  3. 3Create an Outbound Voice Profile
  4. 4Purchase or port phone numbers
  5. 5Configure your PBX with sip.aivo.bz