Common RTP (Audio) Issues

One Way Audio

• No Inbound Audio

  • This usually happens when the PBX advertises either the wrong External Address or it’s Internal Address as it’s External. Causing one side of the audio path to get misrouted.

    • If your PBX has the option Verify it’s advertising the correct External IP or FQDN for your PBX and disable any SIP ALG settings on your Router.

    • If your PBX doesn’t have the ability to advertise a different External IP, try enabling SIP ALG on your Router.

• No Outbound Audio

  • This is usually due to a firewall blocking outbound traffic.

No Audio

• This is usually a firewall issue, make sure your firewall isn’t blocking the RTP Audio traffic. This traffic needs to be allowed from anywhere.

• Some firewalls will track the SIP traffic and automatically allow the RTP as Established or Related traffic, however some firewalls will require you to explicitly allow the RTP traffic.

  • With Asterisk based (i.e. FreePBX) Systems the default is UDP Ports 10,000-20,000

Audio Quality Issues (Garbled, Digitized, or Choppy)

• This usually going to be related to network issues.

• In some rare cases changing Codecs can help

  • If you have a steady (no packer loss and low jitter rate) but slow connection, you can benefit from a high compression codec such as g729. Otherwise high compression codecs will exacerbate network issues as you lose a larger chunk of the audio with each dropped/discard packet.

Note: With VoIP/SIP calls being digital, there is no such thing as "static" on a digital call. The term "Static" typically implies whitenoise that is usually associated with Analog Services such as POTs lines. If a user on a purely digital call with no analog legs (such as a when using ATA gateways) describes an Audio issue as having Static. You’ll need to clarify what they mean. Usually what they mean is Choppy or a Digitized (Robotic) sounding audio. Recordings of these calls can be immensely helpful.