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SIP Troubleshooting: Fix Registration Failed, One-Way Audio and More

Updated 2026-09-30 · By the ZuteSIP team (ZuteSIP Inc.)

Short answer: Most SIP problems have one of three causes: wrong credentials or server (registration failed), a router or firewall rewriting SIP traffic (one-way audio, calls dropping after about 30 seconds), or a codec or DTMF mismatch. Check the username, password and sip.zutesipdialer.com first, then disable SIP ALG, try TCP or TLS, and force G.711 codecs.

SIP uses two separate paths: signalling (registration, ringing, hang-up) and media (the actual voice, carried over RTP). Knowing which path is failing tells you where to look. Use the table below to match your symptom, then follow the detailed fixes.

Quick diagnosis table

SymptomLikely causeFix
"Registration failed" or 401 Unauthorized repeatingWrong username or password, or a space pasted inCopy both again from the number's page; enter the same value in username and auth ID
403 ForbiddenCredentials accepted by format but not valid for this account, or number not activeConfirm the number is active and not in a renewal grace period; recheck credentials
408 Request Timeout or no responseWrong server, blocked port, or DNS problemUse sip.zutesipdialer.com; try TCP 5060 or TLS 5061; test another network
Registered, but incoming calls never ringRegistration expired behind NAT, or app asleepLower re-register interval, enable keep-alive, keep the app running
One-way audioNAT or SIP ALG rewriting addresses; RTP blockedDisable SIP ALG, enable STUN or rport in the app, open outbound UDP for RTP
No audio either wayCodec mismatch or RTP fully blockedAllow only PCMU and PCMA; check firewall and VPN
Call drops after about 30 secondsACK lost because of NAT or ALGDisable SIP ALG, switch to TCP or TLS
488 Not Acceptable HereNo common codecEnable G.711 µ-law and A-law
Keypad presses ignored by menusDTMF mode mismatchSet RFC 2833 (RTP events); try SIP INFO if needed
Choppy or robotic audioPacket loss or jitter on the networkUse wired or stronger Wi-Fi, pause large uploads, close VPN

Fixing "SIP registration failed"

Registration is the step where your app proves who it is. SIP uses digest authentication (defined in IETF RFC 3261): the server replies 401 with a challenge, and the app answers with a hash of your password. If the password is wrong, you see 401 again and again, or eventually 403.

  1. Open the number's page in the ZuteSIP dashboard and copy the SIP username and password again. Paste them; do not retype. Check that no space was added at the end.
  2. Set the server, domain and registrar to sip.zutesipdialer.com. Leave outbound proxy empty or use the same address.
  3. If your app has both "username" and "authorization username", put the SIP username in both.
  4. Check the port: 5060 for UDP or TCP, 5061 for TLS. A TLS account on port 5060, or the reverse, will never register.
  5. Confirm the number is active in your dashboard. If a renewal could not be charged, you are in the 7-day grace period; top up the wallet to keep the number.
  6. Try a different network, such as a mobile hotspot. If it registers there, your original network is blocking or altering SIP.

Fixing one-way audio in SIP

One-way audio means signalling works (the call connects) but voice only travels in one direction. In almost every case, a router is changing the IP addresses inside the SIP message or blocking the return RTP stream. SIP carries the media address inside its body, so if your app advertises a private address such as 192.168.x.x, the far end sends audio to a place it cannot reach.

Many home and office routers have a feature called SIP ALG (Application Layer Gateway) that tries to fix this and often breaks it instead. The IETF discusses the problems such middleboxes cause for SIP in RFC 5853.

  • Disable SIP ALG in the router, often under NAT, Firewall or Advanced settings. Restart the router afterwards.
  • In your SIP app, enable STUN (a standard way to discover your public address, IETF RFC 8489) or "rport" / "use received address" options.
  • Switch transport to TCP 5060 or TLS 5061. Routers usually leave TLS traffic alone because they cannot read it.
  • Allow outbound UDP for RTP. Strict firewalls that only allow web traffic block voice.
  • Turn off VPNs while testing; some drop UDP or add enough delay to break audio.

Calls dropping after about 30 seconds

A call that connects and then ends around the 30-second mark is a classic SIP pattern. The server sends a confirmation and waits for your app's ACK. If the ACK never arrives because a router sent it to the wrong place, the server retransmits for about 32 seconds, as set by the SIP timers in RFC 3261, and then gives up. The fixes are the same as for one-way audio: disable SIP ALG and try TCP or TLS.

Codec and DTMF problems

ZuteSIP uses G.711 µ-law (PCMU) and A-law (PCMA), the ITU-T G.711 codec carried as RTP payload types 0 and 8 (IETF RFC 3551). If your app or PBX offers only compressed codecs such as G.729 or Opus, there is no common codec and the call fails with 488 or connects silently. Enable PCMU and PCMA and move them to the top.

For key presses, use RFC 2833 (now RFC 4733) RTP events. If an automated menu ignores your digits, check that the app is not sending in-band audio tones, and try SIP INFO as the alternative.

Codec order:   PCMU (G.711 u-law), PCMA (G.711 A-law)
Disabled:      G.729, G.722, Opus, GSM, iLBC
DTMF:          RFC 2833 / RFC 4733 (fallback: SIP INFO)
Transport:     UDP 5060  ->  TCP 5060  ->  TLS 5061
Re-register:   300 seconds or less behind NAT

Incoming calls not ringing

If you can call out but incoming calls do not reach your app, the registration has usually lapsed on the router side. Routers forget idle connections after a while, and then the server cannot reach your app. Lower the re-registration interval (for example to 300 seconds or less), enable keep-alive packets, and on mobile allow the app to run in the background.

As a fallback, set call forwarding in the dashboard to send incoming calls to any phone, so you never miss a call while you fix the SIP side.

When to contact support

If you have worked through the table and still see errors, open a support ticket from the dashboard or email support@zutesipdialer.com. Include the number, the app and version, the transport and port, the exact error code, and the time of a failed attempt. Never send your SIP password; support does not need it.

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Frequently asked questions

Why does my SIP app say registration failed?

The most common causes are a wrong or mistyped password, the wrong server, or a blocked port. Recopy the credentials from the number's page, use sip.zutesipdialer.com, and try TCP 5060 or TLS 5061.

What causes one-way audio on SIP calls?

Usually NAT: a router or its SIP ALG rewrites addresses so audio is sent to the wrong place, or the firewall blocks the return RTP stream. Disable SIP ALG, enable STUN or rport, and try TLS.

Why do my SIP calls drop after 30 seconds?

The server did not receive your app's ACK, typically because of NAT or SIP ALG, and stopped retrying after the standard timeout. The fix is the same as for one-way audio.

Which codecs should I enable?

Enable G.711 µ-law (PCMU) and A-law (PCMA) only, and put them first. Other codecs will not negotiate.

My number is registered but incoming calls do not ring. What now?

Lower the re-registration interval, enable keep-alive and keep the app running. Call forwarding to any phone is a reliable backup meanwhile.

Should I send my SIP password to support?

No. Support needs the number, the app, the error and the time of the attempt, never the password.

Does a VPN cause SIP problems?

It can. Some VPNs block UDP or add delay, which leads to failed registration or broken audio, so test without the VPN.

Related guides and use cases

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