Guidelines for successful VoIP implementation

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

  • 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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