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BrightPBX

Troubleshooting one way audio

One-way audio is usually a client-side issue

Codecs

  • We recommend setting codec preference as g711u, g729, g722.
  • If you have low bandwidth, set g729 as first.
  • Never set g722 as first priority, since it will cause our switch to have to transcode from g722 to g711 when handling PSTN calls
  • In order to guarantee best quality, pay attention that devices use RTP packetization time or ptime of 20ms. It is usually 20ms set by default on most devices.

Firewalls

Many routers/modems come with configurations that can disrupt SIP and RTP traffic. This can lead to the following problems:

  • Phone calls connect, but audio is lost in one or both directions
  • Inbound calls work but outbound calls do not, or vice versa
  • Calls disconnected after 5 - 35 seconds
  • Calls disconnected after 15/30 minutes
  • Incoming calls go directly to voicemail without ringing the phone.
  • Even though you’ve hung up the call, the phone on the other side continues to ring
  • You can dial a phone number, but don’t hear a dial and/or ringing tone.
  • The ATA does not register itself or loses the registration after some time.

Below are possible reasons why the problems occur:

  • Enabled UPnP
  • Enabled SPI (Stateful Packet Inspection) and/or SIP ALG (Application Layer Gateway) that could overwrite/modify SIP headers
  • Enabled other NAT Traversal technique that might overwrite/modify SIP headers
  • Too short UDP timeout of the Firewall – should be at least 120 seconds
  • Disabled Consistent NAT
  • Blocked SIP and RTP ports

If the problem persists, after adjusting the reasons above, try additional steps:

  • If everything indicates successful configuration, please reboot your cable modem and phone adapter
  • Test with a different phone installed on a computer or smartphone like Zoiper
  • Test your router/phone over a different internet connection.
  • If you are using multiple routers and other networking equipment, try plugging your Phone Adapter directly into the Internet modem. If this resolves your issue, you may have to troubleshoot your other devices.
  • Do a factory reset and reconfigure from scratch
  • Upgrade router firmware, if there is bug with SIP handling, latest version could have fix it
  • Try disabling your firewall briefly. Reboot your router and VoIP device and check if you can make/receive calls. If you can, then enable FW again and try the next two solutions.
  • Configure and plugin phone in the router DMZ port.
  • Or forward SIP and RTP Ports to your phone

Issues due to your local network and its configuration may appear intermittently and unpredictably, sometimes only showing up after hours or days of problem free operation. You may find that powering your router and phones off and on may temporarily solve issues. If your router or computer is using NAT (Network Address Translation) or a firewall, these features might close SIP and RTP ports so that packets never reach your phone. When the proper ports are not forwarded or opened, your calls could drop altogether or fail to initiate. For more information regarding this please refer to our NAT and Firewall Traversal recommendations section.

Best practice router/modem settings

1. Make Sure You Don’t Have A “Double NAT” Network

“Double NAT” (network address translation) is a situation where two routers are placed one after another. This can be a common scenario if, for example, your Internet Service Provider (ISP) provides a modem/router combination device, but you’ve also added a wireless router of your own. Although a computer behind a double NAT setup will usually work fairly well, a VoIP phone will exhibit unusual behaviour.

To resolve this, you’ll need to put your modem/router device into “bridged” mode—most models will allow this. Doing so will disable all routing (NAT) functions on the modem/router, so that only your standalone router will perform these functions. Contact your ISP if you need specific guidance.

You could also choose to purchase a standalone modem, to avoid dealing with your ISP’s hardware all together.

2. Disable the SIP Application Layer Gateway (ALG)

Originally designed to be a security component to help certain types of traffic get through your router, most SIP ALGs actually make things worse.

Not all routers have SIP ALG enabled if they even have it to begin with, but those that do will interfere with VoIP traffic, often preventing traffic from reaching your phone correctly or at all.

We have seen all of the following behaviours:

  • Your phone might not get dial tone
  • You could drop out in the middle of a call
  • Your phone might not always ring on incoming calls
  • The wrong phone might ring (if you have more than one)
  • Your phone might continue to ring after being answered

If you have a router that doesn’t let you disable SIP ALG, consider purchasing a different router that either doesn’t have SIP ALG, or lets you disable it.

If replacing your router isn’t a viable option, you could put one phone in your router’s DMZ. Although this will expose your phone to the open Internet, it eliminates the SIP ALG issue without requiring you to purchase new hardware.

Most routers include a firewall, which is a filter that blocks traffic that should not be allowed in or out. For smaller offices with off-the-shelf routers, these firewalls may need to be modified or turned off. On enterprise-level equipment, certain rules may need to be added. The following settings listed below can typically be found in a commercially available router’s web interface and can be easily checked on/off…

3. Ports

Normally you should not need to “open up” any ports or IP address ranges for BrightPBX. The phone, the device behind the firewall makes the connection from inside the network out to BrightPBX. We simply respond to those packets.

In case you have issues to ensure that traffic is not being blocked, the following ports used for VoIP communication to our servers should be opened:

  • SIP Ports 5060, 6060
  • RTP Ports 10000 – 49999

4. Other Router/Firewall Settings

  • Make sure BrightPBX IPs are set as a trusted source.
  • Increase UDP timeout settings up to 120
  • Disable SIP Transformations (SIP ALG, STUN, ICE, SPI etc.)
  • Enable consistent NAT