Troubleshooting one way audio

One-way audio is usually a client-side issue.

Codecs

• We recommend a codec preference of g711u, g729, g722.
• If you have low bandwidth, set g729 first.
• Never set g722 as first priority — it forces our switch to transcode from g722 to g711 when handling PSTN calls.
• For best quality, ensure devices use an RTP packetization time (ptime) of 20 ms. This is the default on most devices.

See How do I configure audio codecs for a device? for where to set this.

Firewalls

Many routers and modems ship with configurations that disrupt SIP and RTP traffic, which leads to symptoms such as:

• Calls connect, but audio is lost in one or both directions
• Inbound calls work but outbound do not, or vice versa
• Calls disconnect after 5–35 seconds
• Calls disconnect after 15 or 30 minutes
• Incoming calls go straight to voicemail without ringing the phone
• The far-end phone keeps ringing after you have hung up
• You can dial a number but hear no dial or ringing tone
• The ATA does not register, or loses registration after a while

Common causes

• Enabled UPnP
• Enabled SPI (Stateful Packet Inspection) or SIP ALG (Application Layer Gateway), which can overwrite or modify SIP headers
• Another NAT traversal technique that modifies SIP headers
• A UDP timeout that is too short — it should be at least 120 seconds
• Consistent NAT disabled
• Blocked SIP and RTP ports

If the problem persists

• Reboot your cable modem and phone adapter
• Test with a different softphone on a computer or smartphone, such as Zoiper
• Test your router or phone over a different internet connection
• If you use multiple routers and networking equipment, plug the phone adapter directly into the internet modem. If that fixes it, troubleshoot the other devices.
• Factory reset and reconfigure the device from scratch
• Upgrade router firmware — a SIP handling bug may already be fixed in a newer version
• Briefly disable your firewall, reboot the router and VoIP device, and test calls. If they work, re-enable the firewall and try the next two suggestions.
• Configure and plug the phone into the router’s DMZ port
• Forward SIP and RTP ports to your phone

Intermittent problems are still configuration problems

Local network issues can appear unpredictably, sometimes only after hours or days of trouble-free operation. Power-cycling the router and phones may temporarily mask them. If your router or computer uses NAT or a firewall, these can close SIP and RTP ports so packets never reach your phone — dropping calls or preventing them from starting.

Best practice router and modem settings

Avoid a double NAT network

“Double NAT” is where two routers sit one after another — common when your ISP provides a modem/router combination and you have added your own wireless router. A computer behind a double NAT usually works fine, but a VoIP phone will behave strangely.

To resolve it, put your modem/router into bridged mode, which disables its routing and NAT functions so only your standalone router performs them. Most models allow this; contact your ISP if you need specific guidance. You could also buy a standalone modem and avoid your ISP’s hardware entirely.

Disable the SIP Application Layer Gateway (ALG)

Originally designed to help certain traffic through your router, most SIP ALGs make things worse. Not every router has one, but those that do will interfere with VoIP traffic. We have seen all of the following:

• The phone gets no dial tone
• Calls drop out mid-conversation
• The phone does not always ring on incoming calls
• The wrong phone rings, where there is more than one
• The phone continues ringing after being answered

If your router does not let you disable SIP ALG, consider a different router that either lacks it or lets you turn it off. If replacing the router is not viable, you could put one phone in the router’s DMZ — this exposes the phone to the open internet, but eliminates the SIP ALG issue without new hardware.

See Detecting and disabling SIP ALG for how to check.

Ports

Normally you should not need to open any ports or IP ranges for BrightPBX. The phone behind the firewall makes the connection outbound to us, and we simply respond to those packets.

If you are having issues and want to confirm traffic is not being blocked, these are the ports used:

Traffic	Ports
SIP	5060, 6060
RTP	10000–49999

Other router and firewall settings

• Set BrightPBX IPs as a trusted source
• Increase UDP timeout settings to 120 seconds
• Disable SIP transformations (SIP ALG, STUN, ICE, SPI and similar)
• Enable consistent NAT

Related

• Best practices to set up a router and/or firewall
• Asymmetric routing and VoIP