SIP Network Configuration
SIP signaling addresses, media IP ranges, ports, codecs, DNS records, and firewall configuration.
SIP Signaling Addresses
AIVO Connect uses these SIP signaling endpoints. Each region has a primary and secondary IP for redundancy.
| Region | FQDN | Primary IP | Secondary IP |
|---|---|---|---|
| US | sip.aivo.bz | 192.76.120.10 | 64.16.250.10 |
| Europe | sip-eu.aivo.bz | 185.246.41.140 | 185.246.41.141 |
| Australia | sip-au.aivo.bz | 103.115.244.145 | 103.115.244.146 |
| Canada | sip-ca.aivo.bz | 192.76.120.31 | 64.16.250.13 |
Use sip.aivo.bz as your default. Use the regional endpoints if you need traffic to stay in a specific region.
Note: 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 the primary and secondary IPs.
DNS Records
All SIP FQDNs support A, SRV, and NAPTR DNS record types.
For DNS-based SIP routing, configure SRV records:
_sip._udp.yourdomain.com 86400 IN SRV 10 10 5060 sip.aivo.bz
_sip._tcp.yourdomain.com 86400 IN SRV 10 10 5060 sip.aivo.bz
_sips._tcp.yourdomain.com 86400 IN SRV 10 10 5061 sip.aivo.bzYour PBX queries the SRV record and routes SIP traffic to the correct endpoint.
Transport Protocols
| Protocol | Port |
|---|---|
| UDP | 5060 |
| TCP | 5060 |
| TLS | 5061 |
Media (RTP)
RTP port range: 16384 to 32768 (UDP).
Firewall Whitelist Subnets
Whitelist the following CIDR ranges for RTP media traffic:
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
103.115.244.128/25
185.246.41.128/25Note: For the current, up-to-date list, go to AIVO Connect > Network Info in your dashboard or visit aivo.bz/sip-info. IP ranges may be updated periodically.
Codecs
Supported codecs:
- G.722
- G.711U (PCMU)
- G.711A (PCMA)
- G.729
- Opus — Supported for inbound and outbound. Inbound requires TLS or TCP transport.
- H.264
We support ptime:20 for RTP.
Firewall Configuration
Recommended Firewall Rules
- Allow outbound SIP:
- Protocol: UDP + TCP
- Destination: sip.aivo.bz (and regional endpoints)
- Port: 5060 (or 5061 for TLS)
- Allow outbound RTP:
- Protocol: UDP
- Destination: AIVO media subnets (see list above)
- Port: 16384-32768
- Allow return traffic (if not using stateful firewall):
- Protocol: UDP + TCP
- Source: AIVO signaling and media IPs
- Port: 5060 + 16384-32768
Disable SIP ALG
SIP ALG (Application Layer Gateway) is a feature on many consumer and small-business routers that modifies SIP packets. It almost always causes problems:
- Registration failures
- One-way audio
- Dropped calls
- Random disconnections
Disable SIP ALG in your router settings. The exact location varies by router brand:
- Netgear: Advanced > WAN Setup > Disable SIP ALG
- Linksys: Administration > Management > Disable SIP ALG
- Ubiquiti/UniFi: Settings > Threat Management > Disable SIP ALG
QoS (Quality of Service)
For best call quality, prioritize SIP and RTP traffic:
- Mark SIP packets as DSCP CS3 (or EF).
- Mark RTP packets as DSCP EF (Expedited Forwarding).
- Configure your router to prioritize these markings.
Quick Start
- Log in to your AIVO Connect dashboard
- Create a SIP Connection (credential, IP, or FQDN auth)
- Create an Outbound Voice Profile
- Purchase or port phone numbers
- Configure your PBX with sip.aivo.bz
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