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Troubleshooting SIP: Registration Errors, One-Way Audio and Beyond

Updated 2026-09-30 · By the SMS Conecta team (SMSConecta Inc.)

Short answer: Nearly every SIP issue traces back to one of three causes: incorrect login details or server address (registration failed), a router or firewall mangling SIP traffic (one-way audio, calls cutting off near 30 seconds), or a codec/DTMF mismatch. Start by double-checking the username, password and sip.smsconecta.com, then turn off SIP ALG, test TCP or TLS, and force G.711 codecs.

SIP relies on two distinct channels — signalling (registration, ringing, hanging up) and media (the actual voice audio, carried over RTP). Figuring out which channel is broken points you to the right fix. Match your symptom against the table below, then work through the detailed steps that follow.

Fast symptom-to-fix 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.smsconecta.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

Resolving a "SIP registration failed" error

Registration is the point where your app proves its identity. SIP relies on digest authentication (laid out in IETF RFC 3261): the server answers with a 401 challenge, and the app responds with a hash built from your password. Get the password wrong and you'll keep seeing 401, or eventually 403.

  1. Go back to the number's page in the SMS Conecta dashboard and re-copy the SIP username and password. Paste them rather than typing them manually, and watch for a stray trailing space.
  2. Point the server, domain and registrar fields all at sip.smsconecta.com. Either leave the outbound proxy blank or set it to the same address.
  3. Where your app lists a separate "authorization username" field alongside "username", put the SIP username in both.
  4. Double-check the port — 5060 for UDP or TCP, 5061 for TLS. Mixing these up (a TLS account on 5060, say) means registration never succeeds.
  5. Make sure the number shows as active in the dashboard. A failed renewal charge puts it into a 7-day grace period — top up the wallet to hold onto the number.
  6. Test from a different network, a mobile hotspot for instance. Registering fine there points to your original network blocking or interfering with SIP.

Solving one-way audio over SIP

One-way audio means signalling is fine — the call connects — but voice only flows in a single direction. Nearly always, a router is rewriting IP addresses inside the SIP message, or blocking the RTP stream coming back. Since SIP embeds the media address inside its own body, an app advertising a private address like 192.168.x.x causes the other side to send audio to a destination it simply cannot reach.

A lot of home and office routers include a "SIP ALG" (Application Layer Gateway) feature meant to fix exactly this, which in practice usually breaks it instead. The IETF covers the trouble these middleboxes cause for SIP in RFC 5853.

  • Turn off SIP ALG in the router — look under NAT, Firewall or Advanced settings — then restart the router.
  • In the SIP app, switch on STUN (the standard method for discovering your public address, per IETF RFC 8489) or enable "rport" / "use received address".
  • Change transport to TCP 5060 or TLS 5061. Routers generally leave TLS traffic untouched since they can't read inside it.
  • Permit outbound UDP for RTP — strict firewalls limited to web traffic will otherwise block voice entirely.
  • Disable any VPN while testing, since some drop UDP packets or introduce enough delay to break audio.

Calls cutting off around the 30-second mark

A call that connects fine and then dies near 30 seconds is a textbook SIP symptom. The server issues a confirmation and sits waiting for your app's ACK. When that ACK never arrives — because a router routed it to the wrong address — the server keeps retransmitting for roughly 32 seconds, per the SIP timers defined in RFC 3261, before giving up entirely. The remedy matches the one-way audio fix: disable SIP ALG and switch to TCP or TLS.

Codec and DTMF troubles

SMS Conecta relies on G.711 µ-law (PCMU) and A-law (PCMA) — the ITU-T G.711 codec, carried as RTP payload types 0 and 8 under IETF RFC 3551. When an app or PBX only offers compressed codecs such as G.729 or Opus, there is nothing in common to negotiate, so the call either fails with 488 or connects with no sound. Switch on PCMU and PCMA and push them to the top of the list.

For key-press signalling, use RFC 2833 (superseded by RFC 4733) RTP events. If an automated phone menu ignores your digits, confirm the app isn't sending in-band audio tones instead, and try SIP INFO as a fallback.

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

When incoming calls stop ringing

Being able to call out while incoming calls never reach your app usually means the registration has lapsed on the router end. Routers drop idle connections over time, after which the server has no path back to your app. Shorten the re-registration interval (300 seconds or less, say), turn on keep-alive packets, and on mobile, let the app keep running in the background.

As a safety net, enable call forwarding in the dashboard so incoming calls land on any phone — that way you won't miss anything while sorting out the SIP side.

Reaching out to support

Still stuck after working through the table above? Open a ticket from the dashboard or email support@smsconecta.com. Mention the number, the app and its version, the transport and port used, the exact error code, and when the failed attempt happened. Never include your SIP password — support never needs it.

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Frequently asked questions

Why does my SIP app report registration failed?

Usually a mistyped or incorrect password, the wrong server address, or a blocked port. Re-copy the credentials straight from the number's page, confirm you're using sip.smsconecta.com, and try TCP 5060 or TLS 5061.

What is behind one-way audio on a SIP call?

NAT is the usual suspect: a router's SIP ALG rewrites addresses so audio ends up misdirected, or a firewall blocks the returning RTP stream. Turn off SIP ALG, enable STUN or rport, and give TLS a try.

Why do my calls keep dropping around 30 seconds in?

The server never received your app's ACK — typically a NAT or SIP ALG issue — and stopped retrying once the standard timeout passed. Fix it the same way as one-way audio.

Which codecs ought to be enabled?

Stick to G.711 µ-law (PCMU) and A-law (PCMA), ranked first in the list. Anything else simply won't negotiate.

My number registers fine but incoming calls stay silent — what now?

Shorten the re-registration interval, turn on keep-alive, and keep the app running continuously. Call forwarding to any phone works as a dependable backup in the meantime.

Is it okay to share my SIP password with support?

No — support only needs the number, the app, the error, and when it happened. The password should never be shared.

Can a VPN be the source of SIP issues?

Yes, potentially. Certain VPNs block UDP or add noticeable delay, leading to failed registration or broken audio — test with it switched off.

Related guides and use cases

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