mirror of
https://github.com/jambonz/sbc-inbound.git
synced 2026-10-05 03:13:23 +00:00
initial changes for handling SIP INFO from webrtc clients
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+26
-2
@@ -3,7 +3,6 @@ const SrsClient = require('@jambonz/siprec-client-utils');
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const {makeRtpEngineOpts, SdpWantsSrtp, makeCallCountKey} = require('./utils');
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const {forwardInDialogRequests} = require('drachtio-fn-b2b-sugar');
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const {parseUri, stringifyUri, SipError} = require('drachtio-srf');
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const { v4: uuidv4 } = require('uuid');
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const debug = require('debug')('jambonz:sbc-inbound');
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const MS_TEAMS_USER_AGENT = 'Microsoft.PSTNHub.SIPProxy';
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const MS_TEAMS_SIP_ENDPOINT = 'sip.pstnhub.microsoft.com';
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@@ -20,7 +19,7 @@ const createBLegFromHeader = (req) => {
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};
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const createSiprecBody = (headers, sdp, type, content) => {
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const sep = uuidv4();
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const sep = 'uniqueBoundary';
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headers['Content-Type'] = `multipart/mixed;boundary="${sep}"`;
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return `--${sep}\r
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Content-Type: application/sdp\r
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@@ -80,6 +79,7 @@ class CallSession extends Emitter {
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unblockMedia,
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blockDTMF,
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unblockDTMF,
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playDTMF,
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subscribeDTMF,
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unsubscribeDTMF,
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subscribeRequest,
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@@ -93,6 +93,7 @@ class CallSession extends Emitter {
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this.unblockMedia = unblockMedia;
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this.blockDTMF = blockDTMF;
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this.unblockDTMF = unblockDTMF;
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this.playDTMF = playDTMF;
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this.subscribeDTMF = subscribeDTMF;
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this.unsubscribeDTMF = unsubscribeDTMF;
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this.subscribeRequest = subscribeRequest;
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@@ -512,6 +513,7 @@ Duration=${payload.duration} `
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async _onInfo(dlg, req, res) {
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const fromTag = dlg.type === 'uas' ? this.rtpEngineOpts.uas.tag : this.rtpEngineOpts.uac.tag;
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const toTag = dlg.type === 'uas' ? this.rtpEngineOpts.uac.tag : this.rtpEngineOpts.uas.tag;
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const contentType = req.get('Content-Type');
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try {
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if (dlg.type === 'uac' && req.has('X-Reason')) {
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const reason = req.get('X-Reason');
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@@ -617,6 +619,28 @@ Duration=${payload.duration} `
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res.send(succeeded ? 200 : 503);
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}
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}
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else if (dlg.type === 'uas' && ['application/dtmf-relay', 'application/dtmf'].includes(contentType)) {
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/* convert SIP INFO dtmf to RFC 2833 */
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const arr = /Signal=\s([1-9#*])/.exec(req.body);
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if (!arr) throw new Error(`_onInfo: no dtmf in body for ${contentType}`);
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const digit = arr[1];
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const arr2 = /Duration=\s(\d+)/.exec(req.body);
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const duration = arr2 ? arr2[1] : 250;
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this.logger.info({body: req.body}, 'got SIP INFO DTMF from caller');
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const opts = {
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...this.rtpEngineOpts.common,
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'from-tag': this.rtpEngineOpts.uas.tag,
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digit,
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duration
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};
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const response = await this.playDTMF(opts);
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this.logger.debug({opts, response}, 'response from rtpengine to playDTMF');
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if ('ok' !== response.result) {
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this.logger.error({response}, `rtpengine playDTMF failed with ${JSON.stringify(response)}`);
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throw new Error('rtpengine failed: answer');
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}
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res.send(200);
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}
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else {
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const immutableHdrs = ['via', 'from', 'to', 'call-id', 'cseq', 'max-forwards', 'content-length'];
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const headers = {};
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+8
-2
@@ -15,7 +15,13 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
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const from = req.getParsedHeader('from');
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const srtpOpts = teams ? srtpCharacteristics['teams'] : srtpCharacteristics['default'];
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const dstOpts = dstIsUsingSrtp ? srtpOpts : rtpCharacteristics;
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const srctOpts = srcIsUsingSrtp ? srtpOpts : rtpCharacteristics;
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const srcOpts = srcIsUsingSrtp ? srtpOpts : rtpCharacteristics;
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/* webrtc clients (e.g. sipjs) send DMTF via SIP INFO */
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if ((srcIsUsingSrtp || dstIsUsingSrtp) && !teams) {
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dstOpts.flags.push('inject DTMF');
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srcOpts.flags.push('inject DTMF');
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}
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const common = {
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'call-id': req.get('Call-ID'),
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'replace': ['origin', 'session-connection']
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@@ -24,7 +30,7 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
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common,
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uas: {
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tag: from.params.tag,
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mediaOpts: srctOpts
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mediaOpts: srcOpts
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},
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uac: {
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tag: null,
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