initial changes for handling SIP INFO from webrtc clients

This commit is contained in:
Dave Horton
2022-08-11 11:08:39 +02:00
parent a94f25b0bd
commit 8663b03b41
2 changed files with 34 additions and 4 deletions
+26 -2
View File
@@ -3,7 +3,6 @@ const SrsClient = require('@jambonz/siprec-client-utils');
const {makeRtpEngineOpts, SdpWantsSrtp, makeCallCountKey} = require('./utils');
const {forwardInDialogRequests} = require('drachtio-fn-b2b-sugar');
const {parseUri, stringifyUri, SipError} = require('drachtio-srf');
const { v4: uuidv4 } = require('uuid');
const debug = require('debug')('jambonz:sbc-inbound');
const MS_TEAMS_USER_AGENT = 'Microsoft.PSTNHub.SIPProxy';
const MS_TEAMS_SIP_ENDPOINT = 'sip.pstnhub.microsoft.com';
@@ -20,7 +19,7 @@ const createBLegFromHeader = (req) => {
};
const createSiprecBody = (headers, sdp, type, content) => {
const sep = uuidv4();
const sep = 'uniqueBoundary';
headers['Content-Type'] = `multipart/mixed;boundary="${sep}"`;
return `--${sep}\r
Content-Type: application/sdp\r
@@ -80,6 +79,7 @@ class CallSession extends Emitter {
unblockMedia,
blockDTMF,
unblockDTMF,
playDTMF,
subscribeDTMF,
unsubscribeDTMF,
subscribeRequest,
@@ -93,6 +93,7 @@ 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;
@@ -512,6 +513,7 @@ Duration=${payload.duration} `
async _onInfo(dlg, req, res) {
const fromTag = dlg.type === 'uas' ? this.rtpEngineOpts.uas.tag : this.rtpEngineOpts.uac.tag;
const toTag = dlg.type === 'uas' ? this.rtpEngineOpts.uac.tag : this.rtpEngineOpts.uas.tag;
const contentType = req.get('Content-Type');
try {
if (dlg.type === 'uac' && req.has('X-Reason')) {
const reason = req.get('X-Reason');
@@ -617,6 +619,28 @@ Duration=${payload.duration} `
res.send(succeeded ? 200 : 503);
}
}
else if (dlg.type === 'uas' && ['application/dtmf-relay', 'application/dtmf'].includes(contentType)) {
/* convert SIP INFO dtmf to RFC 2833 */
const arr = /Signal=\s([1-9#*])/.exec(req.body);
if (!arr) throw new Error(`_onInfo: no dtmf in body for ${contentType}`);
const digit = arr[1];
const arr2 = /Duration=\s(\d+)/.exec(req.body);
const duration = arr2 ? arr2[1] : 250;
this.logger.info({body: req.body}, 'got SIP INFO DTMF from caller');
const opts = {
...this.rtpEngineOpts.common,
'from-tag': this.rtpEngineOpts.uas.tag,
digit,
duration
};
const response = await this.playDTMF(opts);
this.logger.debug({opts, response}, 'response from rtpengine to playDTMF');
if ('ok' !== response.result) {
this.logger.error({response}, `rtpengine playDTMF failed with ${JSON.stringify(response)}`);
throw new Error('rtpengine failed: answer');
}
res.send(200);
}
else {
const immutableHdrs = ['via', 'from', 'to', 'call-id', 'cseq', 'max-forwards', 'content-length'];
const headers = {};
+8 -2
View File
@@ -15,7 +15,13 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
const from = req.getParsedHeader('from');
const srtpOpts = teams ? srtpCharacteristics['teams'] : srtpCharacteristics['default'];
const dstOpts = dstIsUsingSrtp ? srtpOpts : rtpCharacteristics;
const srctOpts = srcIsUsingSrtp ? 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']
@@ -24,7 +30,7 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
common,
uas: {
tag: from.params.tag,
mediaOpts: srctOpts
mediaOpts: srcOpts
},
uac: {
tag: null,