lingsip

Pure SIP signaling library for Go.

lingsip implements the wire and session layers you need to build a SIP UAS/UAC: message parse/serialize, UDP and TCP/TLS transport, RFC 3261 transactions, dialog identifiers, SDP helpers, digest auth, and common carrier headers (PAI, History-Info, STIR/SHAKEN, Reason, DTMF).

It does not implement RTP media, B2BUA bridging, dialplan, or persistence — those stay in your application.

go get github.com/LingByte/lingsip@latest

import (
    "github.com/LingByte/lingsip/stack"
    "github.com/LingByte/lingsip/transaction"
    "github.com/LingByte/lingsip/uas"
    "github.com/LingByte/lingsip/sdp"
)

Requirements: Go 1.26+

---

Features

Area	What you get
Stack	SIP/2.0 parse & serialize, UDP Endpoint, TCP/TLS stream serving
Transactions	INVITE / non-INVITE client & server txs, CANCEL matching, ACK helpers
UAS	Typed method handlers + AttachWithTransaction wiring
Dialog	Call-ID / tags / early→confirmed matching
SDP	Audio offer parse, BuildAudioAnswer codec intersection, SDES / DTLS attrs
Auth	Digest challenge & verify (RFC 2617 / RFC 3261 §22)
Headers	PAI / Privacy, History-Info + Diversion, STIR/SHAKEN PASSporT, Reason
DTMF	RFC 2833/4733 + SIP INFO digit parsing

---

Packages

Package	Role
stack	Messages, constants, UDP endpoint, TCP/TLS
transaction	RFC 3261 transactions
uas	UAS handlers & server-tx middleware
dialog	Dialog identifiers
sdp	SDP parse / generate / answer builder
session_timer	RFC 4028 Session-Expires
auth	Digest authentication
identity	RFC 3325 PAI / Privacy
historyinfo	RFC 7044 History-Info + Diversion
stir	RFC 8224 STIR/SHAKEN
signaling	RFC 3326 Reason / BYE classes
dtmf	Telephone-event & SIP INFO digits

---

Quick start

Minimal UAS (REGISTER, INVITE→200+SDP, ACK, BYE, CANCEL, MESSAGE; optional digest + TCP):

git clone https://github.com/LingByte/lingsip.git
cd lingsip
go run ./examples/uas-server -listen 0.0.0.0:5060

# optional:
#   -tcp :5060
#   -realm example.com -user alice -pass secret
#   -public-ip 203.0.113.10
#   -rtp-port 10000

Point a softphone at that address, or drive it from the UAC examples:

# terminal 1
go run ./examples/uas-server -listen 127.0.0.1:6050

# terminal 2
go run ./examples/uac-options -target 127.0.0.1:6050
go run ./examples/uac-message -target 127.0.0.1:6050 -body "hello"

Offline demos (no network):

go run ./examples/sdp-answer          # offer → BuildAudioAnswer
go run ./examples/digest-challenge    # Challenge401 + VerifyRequest

---

Example: answer an INVITE with SDP

offer, err := sdp.Parse(req.Body)
// err / nil offer → BuildAudioAnswer falls back to PCMA + telephone-event

body := sdp.BuildAudioAnswer(localIP, rtpPort, offer)
resp, err := uas.NewResponseWithTo(req, stack.StatusOK, stack.ReasonOK,
    body, stack.ContentTypeSDP, toWithLocalTag)

SelectAnswerCodecs intersects the offer with a PCMA → PCMU → G.722 → Opus preference and keeps telephone-event when present.

---

Design notes

• Signaling only. No RTP sockets, codecs, or media engines.
• No global mutable defaults. Configure via structs and context.
• UDP + stream transports. Client txs: RunInviteClient / RunNonInviteClient (UDP) and *Reliable variants (TCP/TLS).
• UAS wiring. Prefer uas.Handlers.AttachWithTransaction so retransmits, CANCEL, and ACK are handled by transaction.Manager.

---

Status

API may still change before v1.0.0. Pin a tagged release in production.

go get github.com/LingByte/lingsip@v0.1.0

---

Development

go test ./...
go build ./examples/...

---

License

See LICENSE.

lingsip is developed by LingByte and used in production by LingEchoX.