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Feat/3pcc invite #187

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merged 4 commits into from
Dec 12, 2024
+90 −27
Merged

Feat/3pcc invite #187

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c21851c
wip
davehorton Dec 12, 2024
e405509
wip
davehorton Dec 12, 2024
0d584a1
wip
davehorton Dec 12, 2024
094a675
add test for late media / 3pcc invite, which should now work
davehorton Dec 12, 2024
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4 changes: 0 additions & 4 deletions app.js
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Original file line number Diff line number Diff line change
Expand Up @@ -244,10 +244,6 @@ srf.invite((req, res) => {
}
return session.replaces(req, res);
}
if (req.locals.sdp === '') {
logger.info('no sdp in invite');
return res.send(488, {headers: {'X-Reason': '3pcc INVITEs without SDP are not currently supported'}});
}
const session = new CallSession(logger, req, res);
session.connect();
});
Expand Down
36 changes: 26 additions & 10 deletions lib/call-session.js
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Original file line number Diff line number Diff line change
Expand Up @@ -108,7 +108,8 @@ class CallSession extends Emitter {

async connect() {
const {sdp} = this.req.locals;
this.logger.info('inbound call accepted for routing');
const is3pcc = this.req.body?.length === 0;
this.logger.info(`inbound ${is3pcc ? '3pcc ' : ''}call accepted for routing`);
const engine = this.getRtpEngine();
if (!engine) {
this.logger.info('No available rtpengines, rejecting call!');
Expand Down Expand Up @@ -183,13 +184,13 @@ class CallSession extends Emitter {
else uri = `${scheme}:${host}`;
this.logger.info(`uri will be: ${uri}, proxy ${proxy}`);

try {
const sendOfferToRtpEngine = async(remoteSdp) => {
const opts = {
...this.rtpEngineOpts.common,
...this.rtpEngineOpts.uac.mediaOpts,
'from-tag': this.rtpEngineOpts.uas.tag,
direction: [isPrivateVoipNetwork(this.req.source_address) ? 'private' : 'public', 'private'],
sdp
sdp: remoteSdp
};
const startAt = process.hrtime();
const response = await this.offer(opts);
Expand All @@ -202,7 +203,11 @@ class CallSession extends Emitter {
this.logger.error({}, `rtpengine offer failed with ${JSON.stringify(response)}`);
throw new Error('rtpengine failed: answer');
}
return response;
};

try {
const response = await sendOfferToRtpEngine(sdp);
let headers = {
'From': createBLegFromHeader(this.req),
'To': this.req.get('To'),
Expand All @@ -212,9 +217,13 @@ class CallSession extends Emitter {
};
if (this.privateSipAddress) headers = {...headers, Contact: `<sip:${this.privateSipAddress}>`};

const spdOfferB = this.siprec && this.xml ?
createSiprecBody(headers, response.sdp, this.xml.type, this.xml.content) :
response.sdp;
let spdOfferB;
if (this.siprec && this.xml) {
spdOfferB = createSiprecBody(headers, response.sdp, this.xml.type, this.xml.content);
}
else if (!is3pcc) {
spdOfferB = response.sdp;
}

if (this.req.locals.carrier) {
Object.assign(headers, {
Expand Down Expand Up @@ -281,7 +290,10 @@ class CallSession extends Emitter {
'-X-Authenticated-User'
],
proxyResponseHeaders: ['all', '-X-Trace-ID'],
localSdpB: spdOfferB,
localSdpB: spdOfferB ? spdOfferB : async(ackBody) => {
const response = await sendOfferToRtpEngine(ackBody);
return response.sdp;
},
localSdpA: async(sdp, res) => {
this.rtpEngineOpts.uac.tag = res.getParsedHeader('To').params.tag;
const opts = {
Expand All @@ -292,8 +304,12 @@ class CallSession extends Emitter {
sdp
};
const startAt = process.hrtime();
const response = await this.answer(opts);
this.logger.debug({response, opts}, 'response from rtpengine to answer');
const aOpts = {
...opts,
...(is3pcc && {direction: ['private', 'public']})
};
const response = await this.answer(aOpts);
this.logger.debug({response, opts: aOpts}, 'response from rtpengine to answer');
const rtt = roundTripTime(startAt);
this.stats.histogram('app.rtpengine.response_time', rtt, [
'direction:inbound', 'command:answer', `rtpengine:${this.rtpengineIp}`]);
Expand All @@ -313,7 +329,7 @@ class CallSession extends Emitter {
}

return response.sdp;
}
},
});

// successfully connected
Expand Down
77 changes: 64 additions & 13 deletions test/scenarios/uac-late-media.xml
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Original file line number Diff line number Diff line change
Expand Up @@ -19,22 +19,21 @@
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->

<scenario name="UAC with late media">
<scenario name="UAC with media">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500">
<![CDATA[

INVITE sip:16173333456@[remote_ip]:[remote_port] SIP/2.0
INVITE sip:+16173333456@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:16173333456@jambonz.org>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-no-3pcc
Content-Type: application/sdp
Subject: uac-late-media
Content-Length: 0

]]>
Expand All @@ -43,27 +42,79 @@
<recv response="100" optional="true">
</recv>

<recv response="488">
<recv response="180" optional="true">
</recv>

<!-- By adding rrs="true" (Record Route Sets), the route sets -->
<!-- are saved and used for following messages sent. Useful to test -->
<!-- against stateful SIP proxies/B2BUAs. -->
<recv response="200" rtd="true" crlf="true">
</recv>

<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send>
<![CDATA[

ACK sip:16173333456@[remote_ip]:[remote_port] SIP/2.0
[last_Via]
ACK sip:16173333456@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
To: <sip:16173333456@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Subject: uac-no-3pcc
Max-Forwards: 70
Subject: uac-late-media
Content-Type: application/sdp
Content-Length: [len]

v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio [auto_media_port] RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16


]]>
</send>

<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>

<!-- Pause briefly -->
<pause milliseconds="3000"/>

<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[

BYE sip:16173333456@jambonz.org SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:16173333456@jambonz.org>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Subject: uac-late-media
Content-Length: 0

]]>
</send>

<!-- definition of the response time repartition table (unit is ms) -->
<recv response="200" crlf="true">
</recv>

<!-- definition of the response time repartition table (unit is ms) -->
<ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

<!-- definition of the call length repartition table (unit is ms) -->
<CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
</scenario>

</scenario>

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