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Allow requesting a ringtone during SIP transfers #207

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Oct 23, 2024
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2 changes: 1 addition & 1 deletion go.mod
Original file line number Diff line number Diff line change
Expand Up @@ -13,7 +13,7 @@ require (
github.com/jfreymuth/oggvorbis v1.0.5
github.com/livekit/mageutil v0.0.0-20230125210925-54e8a70427c1
github.com/livekit/mediatransportutil v0.0.0-20240730083616-559fa5ece598
github.com/livekit/protocol v1.26.1-0.20241016113321-d16f740cf07b
github.com/livekit/protocol v1.26.1-0.20241022031344-538889e5de0a
github.com/livekit/psrpc v0.6.1-0.20240924010758-9f0a4268a3b9
github.com/livekit/server-sdk-go/v2 v2.2.2-0.20241015094126-b8538ae5d67b
github.com/mjibson/go-dsp v0.0.0-20180508042940-11479a337f12
Expand Down
4 changes: 2 additions & 2 deletions go.sum
Original file line number Diff line number Diff line change
Expand Up @@ -118,8 +118,8 @@ github.com/livekit/mageutil v0.0.0-20230125210925-54e8a70427c1 h1:jm09419p0lqTkD
github.com/livekit/mageutil v0.0.0-20230125210925-54e8a70427c1/go.mod h1:Rs3MhFwutWhGwmY1VQsygw28z5bWcnEYmS1OG9OxjOQ=
github.com/livekit/mediatransportutil v0.0.0-20240730083616-559fa5ece598 h1:yLlkHk2feSLHstD9n4VKg7YEBR4rLODTI4WE8gNBEnQ=
github.com/livekit/mediatransportutil v0.0.0-20240730083616-559fa5ece598/go.mod h1:jwKUCmObuiEDH0iiuJHaGMXwRs3RjrB4G6qqgkr/5oE=
github.com/livekit/protocol v1.26.1-0.20241016113321-d16f740cf07b h1:kbGOwqbLMPTw8zMn8vluHoJIR+NjLDWbsMBI8GWJwzQ=
github.com/livekit/protocol v1.26.1-0.20241016113321-d16f740cf07b/go.mod h1:nxRzmQBKSYK64gqr7ABWwt78hvrgiO2wYuCojRYb7Gs=
github.com/livekit/protocol v1.26.1-0.20241022031344-538889e5de0a h1:31YXXJLEwCflp7KEe9rRAwmONyCwHFujTl4MdxegTxw=
github.com/livekit/protocol v1.26.1-0.20241022031344-538889e5de0a/go.mod h1:nxRzmQBKSYK64gqr7ABWwt78hvrgiO2wYuCojRYb7Gs=
github.com/livekit/psrpc v0.6.1-0.20240924010758-9f0a4268a3b9 h1:33oBjGpVD9tYkDXQU42tnHl8eCX9G6PVUToBVuCUyOs=
github.com/livekit/psrpc v0.6.1-0.20240924010758-9f0a4268a3b9/go.mod h1:CQUBSPfYYAaevg1TNCc6/aYsa8DJH4jSRFdCeSZk5u0=
github.com/livekit/server-sdk-go/v2 v2.2.2-0.20241015094126-b8538ae5d67b h1:R1GpKwVbSYsG08k5sIkNCukvnrkOE18R8IO1YeujR8o=
Expand Down
2 changes: 1 addition & 1 deletion pkg/sip/client.go
Original file line number Diff line number Diff line change
Expand Up @@ -187,7 +187,7 @@ func (c *Client) createSIPParticipant(ctx context.Context, req *rpc.InternalCrea
user: req.Username,
pass: req.Password,
dtmf: req.Dtmf,
ringtone: req.PlayRingtone,
dialtone: req.PlayDialtone,
headers: req.Headers,
headersToAttrs: req.HeadersToAttributes,
ringingTimeout: req.RingingTimeout.AsDuration(),
Expand Down
37 changes: 35 additions & 2 deletions pkg/sip/inbound.go
Original file line number Diff line number Diff line change
Expand Up @@ -18,13 +18,15 @@ import (
"context"
"errors"
"fmt"
"math"
"net/netip"
"slices"
"sync"
"sync/atomic"
"time"

"github.com/emiago/sipgo/sip"
"github.com/frostbyte73/core"
"github.com/icholy/digest"

"github.com/livekit/protocol/livekit"
Expand All @@ -39,6 +41,7 @@ import (
"github.com/livekit/sip/pkg/media"
"github.com/livekit/sip/pkg/media/dtmf"
"github.com/livekit/sip/pkg/media/rtp"
"github.com/livekit/sip/pkg/media/tones"
"github.com/livekit/sip/pkg/stats"
"github.com/livekit/sip/res"
)
Expand Down Expand Up @@ -274,6 +277,7 @@ type inboundCall struct {
joinDur func() time.Duration
forwardDTMF atomic.Bool
done atomic.Bool
started core.Fuse
}

func (s *Server) newInboundCall(
Expand Down Expand Up @@ -427,6 +431,9 @@ func (c *inboundCall) handleInvite(ctx context.Context, req *sip.Request, trunkI
return // already sent a response
}
}

c.started.Break()

// Wait for the caller to terminate the call.
select {
case <-ctx.Done():
Expand Down Expand Up @@ -766,8 +773,34 @@ func (c *inboundCall) handleDTMF(tone dtmf.Event) {
}
}

func (c *inboundCall) transferCall(ctx context.Context, transferTo string) error {
err := c.cc.TransferCall(ctx, transferTo)
func (c *inboundCall) transferCall(ctx context.Context, transferTo string, dialtone bool) (retErr error) {
var err error

if dialtone && c.started.IsBroken() && !c.done.Load() {
const ringVolume = math.MaxInt16 / 2
rctx, rcancel := context.WithCancel(ctx)
defer rcancel()

// mute the room audio to the SIP participant
w := c.lkRoom.SwapOutput(nil)
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Since w may contain C resampler, we need to make sure it's either set back via SwapOutput, or closed if it's no longer needed.

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Sorry for confusing, but this still is not closed. The c.media.GetAudioWriter() is fine, it will be closed when c.media is closed. The problem is with w - you detach it from the room, so it won't be closed if the transfer succeeds. It should probably be in the defer below - it either swaps w back to the room, or should close it.


defer func() {
if retErr != nil && !c.done.Load() {
c.lkRoom.SwapOutput(w)
} else {
w.Close()
}
}()

go func() {
aw := c.media.GetAudioWriter()

tones.Play(rctx, aw, ringVolume, tones.ETSIRinging)
aw.Close()
}()
}

err = c.cc.TransferCall(ctx, transferTo)
if err != nil {
c.log.Infow("inbound call failed to transfer", "error", err, "transferTo", transferTo)
return err
Expand Down
53 changes: 39 additions & 14 deletions pkg/sip/outbound.go
Original file line number Diff line number Diff line change
Expand Up @@ -51,7 +51,7 @@ type sipOutboundConfig struct {
user string
pass string
dtmf string
ringtone bool
dialtone bool
headers map[string]string
headersToAttrs map[string]string
ringingTimeout time.Duration
Expand All @@ -63,15 +63,15 @@ type outboundCall struct {
log logger.Logger
cc *sipOutbound
media *MediaPort
started core.Fuse
stopped core.Fuse
closing core.Fuse

mu sync.RWMutex
mon *stats.CallMonitor
lkRoom *Room
lkRoomIn media.PCM16Writer // output to room; OPUS at 48k
sipConf sipOutboundConfig
sipRunning bool
mu sync.RWMutex
mon *stats.CallMonitor
lkRoom *Room
lkRoomIn media.PCM16Writer // output to room; OPUS at 48k
sipConf sipOutboundConfig
}

func (c *Client) newCall(ctx context.Context, conf *config.Config, log logger.Logger, id LocalTag, room RoomConfig, sipConf sipOutboundConfig) (*outboundCall, error) {
Expand Down Expand Up @@ -223,6 +223,7 @@ func (c *outboundCall) ConnectSIP(ctx context.Context) error {
return fmt.Errorf("update SIP failed: %w", err)
}
c.connectMedia()
c.started.Break()
c.lkRoom.Subscribe()
c.log.Infow("Outbound SIP call established")
return nil
Expand All @@ -247,7 +248,7 @@ func (c *outboundCall) connectToRoom(ctx context.Context, lkNew RoomConfig) erro
if err := r.Connect(c.c.conf, lkNew); err != nil {
return err
}
// We have to create the track early because we might play a ringtone while SIP connects.
// We have to create the track early because we might play a dialtone while SIP connects.
// Thus, we are forced to set full sample rate here instead of letting the codec adapt to the SIP source sample rate.
local, err := r.NewParticipantTrack(RoomSampleRate)
if err != nil {
Expand All @@ -260,12 +261,12 @@ func (c *outboundCall) connectToRoom(ctx context.Context, lkNew RoomConfig) erro
}

func (c *outboundCall) dialSIP(ctx context.Context) error {
if c.sipConf.ringtone {
if c.sipConf.dialtone {
const ringVolume = math.MaxInt16 / 2
rctx, rcancel := context.WithCancel(ctx)
defer rcancel()

// Play a ringtone to the room while participant connects
// Play dialtone to the room while participant connects
go func() {
rctx, span := tracer.Start(rctx, "tones.Play")
defer span.End()
Expand All @@ -286,7 +287,6 @@ func (c *outboundCall) dialSIP(ctx context.Context) error {
}
c.setStatus(CallActive)

c.sipRunning = true
return nil
}

Expand Down Expand Up @@ -327,7 +327,6 @@ func sipResponse(ctx context.Context, tx sip.ClientTransaction, stop <-chan stru
func (c *outboundCall) stopSIP(reason string) {
c.mon.CallTerminate(reason)
c.cc.Close()
c.sipRunning = false
}

func (c *outboundCall) setStatus(v CallStatus) {
Expand Down Expand Up @@ -433,8 +432,34 @@ func (c *outboundCall) handleDTMF(ev dtmf.Event) {
}, lksdk.WithDataPublishReliable(true))
}

func (c *outboundCall) transferCall(ctx context.Context, transferTo string) error {
err := c.cc.transferCall(ctx, transferTo)
func (c *outboundCall) transferCall(ctx context.Context, transferTo string, dialtone bool) (retErr error) {
var err error

if dialtone && c.started.IsBroken() && !c.stopped.IsBroken() {
const ringVolume = math.MaxInt16 / 2
rctx, rcancel := context.WithCancel(ctx)
defer rcancel()

// mute the room audio to the SIP participant
w := c.lkRoom.SwapOutput(nil)

defer func() {
if retErr != nil && !c.stopped.IsBroken() {
c.lkRoom.SwapOutput(w)
} else {
w.Close()
}
}()

go func() {
aw := c.media.GetAudioWriter()

tones.Play(rctx, aw, ringVolume, tones.ETSIRinging)
aw.Close()
}()
}

err = c.cc.transferCall(ctx, transferTo)
if err != nil {
c.log.Infow("outound call failed to transfer", "error", err, "transferTo", transferTo)
return err
Expand Down
8 changes: 4 additions & 4 deletions pkg/sip/service.go
Original file line number Diff line number Diff line change
Expand Up @@ -151,7 +151,7 @@ func (s *Service) TransferSIPParticipant(ctx context.Context, req *rpc.InternalT
ctx, cdone := context.WithTimeout(context.WithoutCancel(ctx), 30*time.Second)
defer cdone()

err := s.processParticipantTransfer(ctx, req.SipCallId, req.TransferTo)
err := s.processParticipantTransfer(ctx, req.SipCallId, req.TransferTo, req.PlayDialtone)
transfetResult.Store(&err)
close(done)

Expand All @@ -177,14 +177,14 @@ func (s *Service) TransferSIPParticipant(ctx context.Context, req *rpc.InternalT
}
}

func (s *Service) processParticipantTransfer(ctx context.Context, callID string, transferTo string) error {
func (s *Service) processParticipantTransfer(ctx context.Context, callID string, transferTo string, dialtone bool) error {
// Look for call both in client (outbound) and server (inbound)
s.cli.cmu.Lock()
out := s.cli.activeCalls[LocalTag(callID)]
s.cli.cmu.Unlock()

if out != nil {
err := out.transferCall(ctx, transferTo)
err := out.transferCall(ctx, transferTo, dialtone)
if err != nil {
return err
}
Expand All @@ -197,7 +197,7 @@ func (s *Service) processParticipantTransfer(ctx context.Context, callID string,
s.srv.cmu.Unlock()

if in != nil {
err := in.transferCall(ctx, transferTo)
err := in.transferCall(ctx, transferTo, dialtone)
if err != nil {
return err
}
Expand Down
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