Correcting one-way audio issues with Asterisk

Troubleshooting One-way Audio in VoIP

One-way audio issues are rarely the fault of your terminating call provider, they are an issue with the end-user VoIP phone, internet connection, firewall or their PBX SIP server.

Fixing one-way audio issues in VoIP are best done one step at a time. Start by eliminating any double NAT possibilities by disabling NAT on any secondary routers that may be present on the LAN. This should be considered good basic LAN network design anyway and multiple instances of NAT will cause problems both in VoIP and elsewhere.

One-way audio can occur in either direction, however in-bound audio failure (lack of audio from the outside caller reaching the inside network (LAN) phone) is probably the most common. In many of these cases, routers and firewalls could be the cause of audio not passing. Typically the voice traffic (RTP) is not being passed to the correct SIP port for its journey inside the premise network.

Certain routers have been known to give SIP issues, although many of them have corrected issues with later models and later firmware. SIP ALG or SIP transformations are notorious for causing issues with VoIP and can be disabled on most routers for better VoIP quality.

You’ll find below a link to a guide to where to find these settings in some routers. SIP transformations are known to corrupt some of the SIP headers resulting in issues with the transfer of the voice traffic correctly.

FOLLOW THESE BASIC STEPS TO ISOLATE THE CAUSE OF ONE-WAY AUDIO:

• Connect the ATA/IAD or another SIP device directly to the first device on the LAN such as the modem. Check for normal operation and good two-way audio. If you experience one-way audio or do not receive dial tone, then the issue will most likely be SIP unfriendly NAT or a firewall present that is not allowing the correct VoIP packets to cross.
• Try setting up port forwarding or IP tunnelling. This will send the needed packets directly to the IAD/ATA device. Typically you would forward UDP ports 5060, 5061 and a range of higher UDP ports for the voice RTP transmission, like 10,000-20,000. You should discuss which port ranges are used by your VoIP provider and use what they recommend.
• Placing the IAD/ATA in a DMZ or perimeter zone on the router will prevent it from having it’s packets changed. *
• Place the IAD/ATA outside of any firewall and NAT, giving it a public IP address. *
• Enable and configure QOS (Quality of Service) to prioritize for Voice traffic.

*Note: These options could place the VOIP ATA in a venerable place for unwanted remote access. Use caution if ever leaving your VoIP PBX exposed to easy outside access.

If after trying the above steps you are able to get a good two-way audio voice, then the next step would be to work inwards into your LAN to determine where the voice transmission is being stopped. Once found you should be able to configure the device to allow needed VoIP packets to pass with breaking or changing the correct SIP information.

Some routers/firewalls use SIP transformations. This setting has been known to play havoc with some SIP headers and should be tried both off and on to see which way would be best for your VoIP. Typically we recommend that it be turned off, as with firewalls past experience has shown it to cause problems. Other settings which should be looked at are devices that do packet inspection as this can be the cause for some instances of one-way audio.