How can I avoid problems when implementing VoIP?
Almost all issues occur due to incorrect configuration of infrastructure and/ or hardware. The following items apply as a generic guide and should be observed in all situations, regardless of the hardware being used.
VoIP implementation guidelines
General
- Avoid NAT behind NAT at all times.
- SIP functions (e.g. SIP ALG) in the router must always be switched off.
- Do not use STUN, TURN, ICE, or UPnP.
- Switch on Keep-A-Live (10 seconds is a good value).
- Port Forwarding is not necessary and is not recommended.
- Preferably use a dedicated internet connection for VoIP to rule out interference from other data traffic. Contention ratio (1:1).
- Dimension the data line on the basis of 0.1 Mbps per call (up and down) when using G.711 A-law (PCMA), or 0.033 Mbps per call when using G.729.
- Make sure the MTU in the modem/ router is correctly set for the data line used.
- Use a separate network for voice and data (VLAN).
- Configure QOS in switches and routers to guarantee bandwidth and prioritize voice traffic.
- Always use hostnames for registration. IP addresses may change.
- The proxy domain supports SRV records.
- SIP/ UDP is preferred over SIP/ TCP.
- Preferably use G.711 A-law (PCMA). This gives the best sound quality. Where bandwidth is limited G.729a may be considered.
- DTMF support is based on RFC2833
Hosted PBX
- Preferably use the G.711a (PCMA) codec with a packet size of 20 milliseconds
- Take advantage of a unique local listen port for each SIP device (5061, 5062 etc.)
- A good NAT router is important for hosted VoIP telephony. Good examples: Cisco 800/1800, Mikrotik (Router Board)
Your own IP PBX
- Preferably do not install the IP PBX behind NAT. If it is desired to do so, place the PBX behind NAT 1:1 with an unused public IP address.
- Avoid dynamic NAT. This often causes problems.
- Provide the IP PBX with a firewall that only allows traffic to and from a restricted IP range. More info…
- Let the IP PBX authenticate/ match on account_id and not on hostname. Incoming traffic can come from multiple hosts and therefore can be denied.
- A static trunk is more reliable than a dynamic trunk with registration.
- By default DID/ DDI is set to the full number (including country code with a ‘+’ prefix). This can be changed in Freedom if desired.
- Outgoing CallerID has the same format. (+27210123456)
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