Common RTP Issues

• 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. 

• 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.