livekit-sip BYE to inbound call routing error

[jonathanciechanowski]

BYE routes back to itself instead of reaching the remote side → 481 → caller stuck on dead audio

What's happening

When livekit-sip tries to hang up an inbound call, the BYE it sends includes its own address in the Route headers. The BYE loops through the proxy chain and comes right back to livekit-sip instead of reaching the trunk provider. The call never actually gets torn down on the remote end, so the caller sits on dead audio until the provider's RTP timeout kicks in (~49 seconds in our case).

This looks related to the closed #112, but that fix addressed a missing From tag. This is a different problem — the Route set itself is wrong.

Setup

livekit/sip	v1.2.0 — digest sha256:96ef59d461e6, git 8c4e964, built 2025-11-11
livekit-server-sdk	2.15.0
@livekit/rtc-node	0.12.2
Trunk provider	Telnyx (inbound, UDP)
Topology	Telnyx FreeSWITCH → two SIP proxies (TCP) → livekit-sip (TCP)

Full SIP trace

<details> <summary>Click to expand — complete annotated pcap dump</summary>

# ─── INVITE ────────────────────────────────────────────────────────
# T+0.000s  <PROXY_A>:5060 → <LIVEKIT_SIP>:5060  TCP
INVITE sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060 SIP/2.0

Record-Route: <sip:<PROXY_A>;transport=tcp;r2=on;lr;ftag=<FTAG>>

Record-Route: <sip:<PROXY_B>;r2=on;lr;ftag=<FTAG>>

Via: SIP/2.0/TCP <PROXY_A>;branch=z9hG4bK2128.d1817250b841b56ec747a99ce287860e.0

Via: SIP/2.0/UDP <TELNYX_FS>:6000;received=<TELNYX_FS>;rport=6000;branch=z9hG4bKKNpXFtNcv1SUB

Max-Forwards: 58

From: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>

To: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>

Call-ID: <REDACTED_CALL_ID>

CSeq: 112937604 INVITE

Contact: <sip:mod_sofia@<TELNYX_FS>:6000>

Content-Type: application/sdp

Content-Length: 356
v=0

o=Telnyx 1774963398 1774963399 IN IP4 <TELNYX_MEDIA>

s=Telnyx

c=IN IP4 <TELNYX_MEDIA>

t=0 0

m=audio 19394 RTP/AVP 9 0 8 18 101

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

a=rtcp:19395 IN IP4 <TELNYX_MEDIA>

a=ptime:20
─── 100 Processing ───────────────────────────────────────────────
T+0.044s  <LIVEKIT_SIP>:5060 → <PROXY_A>  TCP
SIP/2.0 100 Processing

Record-Route: <sip:<LIVEKIT_SIP>:5060;transport=tcp;lr>

Record-Route: <sip:<PROXY_A>;lr;ftag=<FTAG>;transport=tcp;r2=on>

Record-Route: <sip:<PROXY_B>;r2=on;lr;ftag=<FTAG>>

Via: SIP/2.0/TCP <PROXY_A>;branch=z9hG4bK2128.d1817250b841b56ec747a99ce287860e.0

Via: SIP/2.0/UDP <TELNYX_FS>:6000;received=<TELNYX_FS>;rport=6000;branch=z9hG4bKKNpXFtNcv1SUB

From: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>

To: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>

Call-ID: <REDACTED_CALL_ID>

CSeq: 112937604 INVITE

Content-Length: 0
─── 180 Ringing ──────────────────────────────────────────────────
T+0.046s  <LIVEKIT_SIP>:5060 → <PROXY_A>  TCP
SIP/2.0 180 Ringing

Record-Route: <sip:<LIVEKIT_SIP>:5060;transport=tcp;lr>

Record-Route: <sip:<PROXY_A>;ftag=<FTAG>;transport=tcp;r2=on;lr>

Record-Route: <sip:<PROXY_B>;r2=on;lr;ftag=<FTAG>>

Via: SIP/2.0/TCP <PROXY_A>;branch=z9hG4bK2128.d1817250b841b56ec747a99ce287860e.0

Via: SIP/2.0/UDP <TELNYX_FS>:6000;branch=z9hG4bKKNpXFtNcv1SUB;received=<TELNYX_FS>;rport=6000

From: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>

To: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>;tag=<LTAG>

Call-ID: <REDACTED_CALL_ID>

CSeq: 112937604 INVITE

Content-Length: 0
─── 200 OK ───────────────────────────────────────────────────────
T+0.560s  <LIVEKIT_SIP>:5060 → <PROXY_A>  TCP
SIP/2.0 200 OK

Record-Route: <sip:<LIVEKIT_SIP>:5060;lr;transport=tcp>

Record-Route: <sip:<PROXY_A>;r2=on;lr;ftag=<FTAG>;transport=tcp>

Record-Route: <sip:<PROXY_B>;ftag=<FTAG>;r2=on;lr>

Via: SIP/2.0/TCP <PROXY_A>;branch=z9hG4bK2128.d1817250b841b56ec747a99ce287860e.0

Via: SIP/2.0/UDP <TELNYX_FS>:6000;received=<TELNYX_FS>;rport=6000;branch=z9hG4bKKNpXFtNcv1SUB

From: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>

To: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>;tag=<LTAG>

Call-ID: <REDACTED_CALL_ID>

CSeq: 112937604 INVITE

Content-Length: 222

Contact: <sip:<LIVEKIT_SIP>:5060;transport=tcp>

Content-Type: application/sdp
v=0

o=- 1774963398 1774963400 IN IP4 <LIVEKIT_SIP>

s=LiveKit

c=IN IP4 <LIVEKIT_SIP>

t=0 0

m=audio 16941 RTP/AVP 9 101

a=rtpmap:9 G722/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv
─── ACK ──────────────────────────────────────────────────────────
T+0.603s  <PROXY_A>:5060 → <LIVEKIT_SIP>:5060  TCP
ACK sip:<LIVEKIT_SIP>:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP <PROXY_A>;branch=z9hG4bK2128.5e49f9c6a60ddb153db90772ebdf108d.0

Via: SIP/2.0/UDP <TELNYX_FS>:6000;received=<TELNYX_FS>;rport=6000;branch=z9hG4bKSBejtvt1aQZBH

Route: <sip:<LIVEKIT_SIP>:5060;lr;transport=tcp>

Max-Forwards: 69

From: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>

To: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>;tag=<LTAG>

Call-ID: <REDACTED_CALL_ID>

CSeq: 112937604 ACK

Contact: <sip:mod_sofia@<TELNYX_FS>:6000>

Content-Length: 0
─── BYE from livekit-sip (BROKEN) ───────────────────────────────
T+41.260s  <LIVEKIT_SIP>:54792 → <PROXY_A>:5060  TCP
BYE sip:mod_sofia@<TELNYX_FS>:6000 SIP/2.0

Route: <sip:<LIVEKIT_SIP>:5060;transport=tcp;lr>          ← BUG

Route: <sip:<PROXY_A>;r2=on;lr;ftag=<FTAG>;transport=tcp>

Route: <sip:<PROXY_B>;ftag=<FTAG>;r2=on;lr>

Via: SIP/2.0/TCP <LIVEKIT_SIP>:5060;branch=z9hG4bK.tsYZQnOE4AsUynnI;alias

Max-Forwards: 70

Call-ID: <REDACTED_CALL_ID>

Content-Length: 0

CSeq: 112937605 BYE

From: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>;tag=<LTAG>

To: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>
─── BYE after proxy hop (looping back) ──────────────────────────
T+41.260s  <PROXY_B>:5060 → <LIVEKIT_SIP>:5060  TCP
BYE sip:mod_sofia@<TELNYX_FS>:6000 SIP/2.0

Route: <sip:<LIVEKIT_SIP>:5060;transport=tcp;lr>

Route: <sip:<PROXY_A>;r2=on;lr;ftag=<FTAG>;transport=tcp>

Route: <sip:<PROXY_B>;ftag=<FTAG>;r2=on;lr>

Via: SIP/2.0/TCP <PROXY_B>;branch=z9hG4bK3128.2e0a389d13590bfbfc645976691f4e0b.0;i=181f8d;conid=14217601

Via: SIP/2.0/TCP <LIVEKIT_SIP>:5060;rport=54792;branch=z9hG4bK.tsYZQnOE4AsUynnI;alias

Max-Forwards: 69

Call-ID: <REDACTED_CALL_ID>

Content-Length: 0

CSeq: 112937605 BYE

From: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>;tag=<LTAG>

To: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>
─── 481 (originated by livekit-sip, not Telnyx) ─────────────────
T+41.282s  <LIVEKIT_SIP>:5060 → <PROXY_A>  TCP
SIP/2.0 481 Call does not exist

Via: SIP/2.0/TCP <PROXY_B>;branch=z9hG4bK3128.2e0a389d13590bfbfc645976691f4e0b.0;i=181f8d;conid=14217601

Via: SIP/2.0/TCP <LIVEKIT_SIP>:5060;rport=54792;branch=z9hG4bK.tsYZQnOE4AsUynnI;alias

From: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>;tag=<LTAG>

To: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>

Call-ID: <REDACTED_CALL_ID>

CSeq: 112937605 BYE

Content-Length: 0
─── 481 forwarded back ──────────────────────────────────────────
T+41.283s  <PROXY_A>:5060 → <LIVEKIT_SIP>:54792  TCP
SIP/2.0 481 Call does not exist

Via: SIP/2.0/TCP <LIVEKIT_SIP>:5060;rport=54792;branch=z9hG4bK.tsYZQnOE4AsUynnI;alias

From: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>;tag=<LTAG>

To: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>

Call-ID: <REDACTED_CALL_ID>

CSeq: 112937605 BYE

Content-Length: 0
─── Telnyx BYE (RTP timeout, ~49s later) ────────────────────────
T+89.988s  <PROXY_A>:5060 → <LIVEKIT_SIP>:5060  TCP
BYE sip:<LIVEKIT_SIP>:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP <PROXY_A>;branch=z9hG4bK3128.312b1fc329dcead654cba63ba17dd9e6.0

Via: SIP/2.0/UDP <TELNYX_FS>:6000;received=<TELNYX_FS>;rport=6000;branch=z9hG4bK7vUDaS8j5v0rp

Route: <sip:<LIVEKIT_SIP>:5060;lr;transport=tcp>

Max-Forwards: 69

From: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>

To: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>;tag=<LTAG>

Call-ID: <REDACTED_CALL_ID>

CSeq: 112937605 BYE

Reason: Q.850;cause=16

Content-Length: 0
─── 481 (dialog already gone) ───────────────────────────────────
T+90.010s  <LIVEKIT_SIP>:5060 → <PROXY_A>  TCP
SIP/2.0 481 Call does not exist

Via: SIP/2.0/TCP <PROXY_A>;branch=z9hG4bK3128.312b1fc329dcead654cba63ba17dd9e6.0

Via: SIP/2.0/UDP <TELNYX_FS>:6000;rport=6000;branch=z9hG4bK7vUDaS8j5v0rp;received=<TELNYX_FS>

From: "<CALLER>" <sip:<CALLER>@sip.telnyx.com>;tag=<FTAG>

To: <sip:<CALLED>@sip.<SCREENER_DOMAIN>:5060>;tag=<LTAG>

Call-ID: <REDACTED_CALL_ID>

CSeq: 112937605 BYE

Content-Length: 0

</details>

Expected behavior

livekit-sip should strip its own URI from the Route set when building outbound in-dialog requests like BYE (per RFC 3261 §12.2.1.1). The BYE should make it all the way to the remote UA and get a clean 200 OK back.

[jonathanciechanowski]

Built the latest main branch as of 4/1/26 13:00 UTC, deployed, and tested with no change in behavior. Internal sip server was first record-route on BYE.

[AsadCodeCraft]

same issue i am facing on self hosted livekit sip.

SIP server dose not send BYE signal when i am removing particiaptant from room also deleted room. still call is active on client side till timeout

[hirekelvin]

BYE's are loose routed end-to-end, these are being hop-to-hop routed, it's being dropped along the way with a 481 and the UA that sent the BYE is assuming it was received and the call was terminated at the other end, this is why the call is still active until the timeout and the far end is getting no audio.