initial changes for sip info dtmf (#46)

This commit is contained in:
Dave Horton
2022-08-11 15:13:07 +02:00
committed by GitHub
parent 5a14aa807a
commit 290723f234
2 changed files with 50 additions and 0 deletions
+44
View File
@@ -69,6 +69,8 @@ class CallSession extends Emitter {
this.performLcr = performLcr;
this.lookupCarrierBySid = lookupCarrierBySid;
this.lookupSipGatewaysByCarrier = lookupSipGatewaysByCarrier;
this._mediaReleased = false;
}
get account_sid() {
@@ -79,6 +81,10 @@ class CallSession extends Emitter {
return this.srf.locals.privateSipAddress;
}
get isMediaReleased() {
return this._mediaReleased;
}
async connect() {
const teams = this.teams = this.req.locals.target === 'teams';
const engine = this.srf.locals.getRtpEngine();
@@ -95,6 +101,7 @@ class CallSession extends Emitter {
unblockMedia,
blockDTMF,
unblockDTMF,
playDTMF,
subscribeDTMF,
unsubscribeDTMF,
subscribeRequest,
@@ -109,6 +116,7 @@ class CallSession extends Emitter {
this.unblockMedia = unblockMedia;
this.blockDTMF = blockDTMF;
this.unblockDTMF = unblockDTMF;
this.playDTMF = playDTMF;
this.subscribeDTMF = subscribeDTMF;
this.unsubscribeDTMF = unsubscribeDTMF;
this.subscribeRequest = subscribeRequest;
@@ -560,6 +568,7 @@ Duration=${payload.duration} `
this.logger.info(`got a reinvite from FS to ${reason}`);
sdp = dlg.other.remote.sdp;
answerMedia.flags = ['asymmetric', 'port latching'];
this._mediaReleased = 'release-media' === reason;
}
else {
sdp = await dlg.other.modify(response.sdp);
@@ -586,6 +595,7 @@ Duration=${payload.duration} `
async _onInfo(dlg, req, res) {
try {
const contentType = req.get('Content-Type');
if (dlg.type === 'uas' && req.has('X-Reason')) {
const toTag = this.rtpEngineOpts.uac.tag;
const reason = req.get('X-Reason');
@@ -688,6 +698,40 @@ Duration=${payload.duration} `
res.send(succeeded ? 200 : 503);
}
}
else if (dlg.type === 'uac' && ['application/dtmf-relay', 'application/dtmf'].includes(contentType)) {
const arr = /Signal=\s*([1-9#*])/.exec(req.body);
if (!arr) {
this.logger.info({body: req.body}, '_onInfo: invalid INFO dtmf request');
throw new Error(`_onInfo: no dtmf in body for ${contentType}`);
}
const code = arr[1];
const arr2 = /Duration=\s*(\d+)/.exec(req.body);
const duration = arr2 ? arr2[1] : 250;
if (this.isMediaReleased) {
/* just relay on to the feature server */
this.logger.info({code, duration}, 'got SIP INFO DTMF from caller, relaying to feature server');
this._onDTMF(dlg.other, {event: code, duration})
.catch((err) => this.logger.info({err}, 'Error relaying DTMF to feature server'));
res.send(200);
}
else {
/* else convert SIP INFO to RFC 2833 telephony events */
this.logger.info({code, duration}, 'got SIP INFO DTMF from caller, converting to RFC 2833');
const opts = {
...this.rtpEngineOpts.common,
'from-tag': this.rtpEngineOpts.uac.tag,
code,
duration
};
const response = await this.playDTMF(opts);
if ('ok' !== response.result) {
this.logger.info({response}, `rtpengine playDTMF failed with ${JSON.stringify(response)}`);
throw new Error('rtpengine failed: answer');
}
res.send(200);
}
}
else {
const response = await dlg.other.request({
method: 'INFO',
+6
View File
@@ -7,6 +7,12 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
const srtpOpts = teams ? srtpCharacteristics['teams'] : srtpCharacteristics['default'];
const dstOpts = dstIsUsingSrtp ? srtpOpts : rtpCharacteristics;
const srcOpts = srcIsUsingSrtp ? srtpOpts : rtpCharacteristics;
/* webrtc clients (e.g. sipjs) send DMTF via SIP INFO */
if ((srcIsUsingSrtp || dstIsUsingSrtp) && !teams) {
dstOpts.flags.push('inject DTMF');
srcOpts.flags.push('inject DTMF');
}
const common = {
'call-id': req.get('Call-ID'),
'replace': ['origin', 'session-connection']