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bugfix #8: maintain Call-ID when sending follow-up INVITE with credentials to endpoint that challenged with 401/407

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This commit is contained in:
Dave Horton
2020-12-13 13:41:47 -05:00
parent 1026bcb9cd
commit 4e5fbbd650
8 changed files with 1011 additions and 164 deletions
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6
app.js
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@@ -26,6 +26,12 @@ const {performLcr, lookupAllTeamsFQDNs, lookupAccountBySipRealm} = require('@jam
connectionLimit: process.env.JAMBONES_MYSQL_CONNECTION_LIMIT || 10
}, logger);
srf.locals.dbHelpers = {performLcr, lookupAllTeamsFQDNs, lookupAccountBySipRealm};
const {createHash, retrieveHash} = require('@jambonz/realtimedb-helpers')({
host: process.env.JAMBONES_REDIS_HOST || 'localhost',
port: process.env.JAMBONES_REDIS_PORT || 6379
}, logger);
srf.locals.realtimeDbHelpers = {createHash, retrieveHash};
const {getRtpEngine} = require('@jambonz/rtpengine-utils')(process.env.JAMBONES_RTPENGINES.split(','), logger, {
emitter: srf.locals.stats
});

28
lib/call-session.js
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@@ -5,6 +5,7 @@ const {SipError} = require('drachtio-srf');
const {parseUri} = require('drachtio-srf');
const debug = require('debug')('jambonz:sbc-outbound');
const makeInviteInProgressKey = (callid) => `sbc-out-iip${callid}`;
/**
* this is to make sure the outgoing From has the number in the incoming From
* and not the incoming PAI
@@ -48,6 +49,7 @@ class CallSession extends Emitter {
}
debug(`got engine: ${JSON.stringify(engine)}`);
const {offer, answer, del} = engine;
const {createHash, retrieveHash} = this.srf.locals.realtimeDbHelpers;
this.offer = offer;
this.answer = answer;
this.del = del;
@@ -127,6 +129,20 @@ class CallSession extends Emitter {
while (uris.length) {
const uri = uris.shift();
const passFailure = 0 === uris.length; // only a single target
if (0 === uris.length) {
try {
const key = makeInviteInProgressKey(this.req.get('Call-ID'));
const obj = await retrieveHash(key);
if (obj.callId && obj.cseq) {
Object.assign(headers, {
'Call-ID': obj.callId,
'CSeq': `${obj.cseq} INVITE`
});
}
} catch (err) {
this.logger.info({err}, 'Error retrieving iip key');
}
}
debug(`sending INVITE to ${uri} via ${proxy})`);
this.logger.info(`sending INVITE to ${uri}`);
try {
@@ -150,6 +166,18 @@ class CallSession extends Emitter {
return response.sdp;
}
}, {
cbRequest: async(err, inv) => {
const opts = {
callId: inv.get('Call-ID'),
cseq: ++inv.getParsedHeader('CSeq').seq
};
try {
const key = makeInviteInProgressKey(this.req.get('Call-ID'));
await createHash(key, opts, 5);
} catch (err) {
this.logger.error({err}, 'Error saving Call-ID/CSeq');
}
},
cbProvisional: (response) => {
if (!earlyMedia && [180, 183].includes(response.status) && response.body) earlyMedia = true;
}

798
package-lock.json generated
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File diff suppressed because it is too large Load Diff

7
package.json
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@@ -28,18 +28,19 @@
},
"dependencies": {
"@jambonz/db-helpers": "^0.5.1",
"@jambonz/realtimedb-helpers": "^0.2.19",
"@jambonz/rtpengine-utils": "^0.1.7",
"@jambonz/stats-collector": "0.1.1",
"debug": "^4.3.1",
"drachtio-fn-b2b-sugar": "^0.0.12",
"drachtio-srf": "^4.4.40",
"drachtio-srf": "^4.4.44",
"jambonz-mw-registrar": "^0.1.3",
"pino": "^6.7.0",
"pino": "^6.8.0",
"rtpengine-client": "^0.1.1"
},
"devDependencies": {
"blue-tape": "^1.0.0",
"eslint": "^7.14.0",
"eslint": "^7.15.0",
"eslint-plugin-promise": "^4.2.1",
"nyc": "^15.1.0",
"tap-spec": "^5.0.0"

10
test/docker-compose-testbed.yaml
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@@ -89,3 +89,13 @@ services:
networks:
sbc-outbound:
ipv4_address: 172.39.0.23
sip-uri-auth:
image: drachtio/sipp:latest
command: sipp -sf /tmp/uas-auth.xml
volumes:
- ./scenarios:/tmp
tty: true
networks:
sbc-outbound:
ipv4_address: 172.39.0.24

167
test/scenarios/uac-sip-uri-auth-success.xml Normal file
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@@ -0,0 +1,167 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp 'uac' scenario with pcap (rtp) play -->
<!-- -->
<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:john@172.39.0.24 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:john@172.39.0.24>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: uac-sip-uri-auth-success
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>
<recv response="100" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send>
<![CDATA[
ACK sip:sip:john@172.39.0.24 SIP/2.0
[last_Via:]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
To: <sip:sip:john@jambonz.org>[peer_tag_param]
[last_Call-ID:]
CSeq: 1 ACK
Subject: uac-sip-uri-auth-success
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
INVITE sip:john@172.39.0.24 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]xx
To: <sip:john@172.39.0.24>
[last_Call-ID:]
CSeq: 2 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
[authentication username=foo password=bar]
Max-Forwards: 70
Subject: uac-sip-uri-auth-success
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>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="200">
</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:john@172.39.0.24 SIP/2.0
[last_Via:]
[last_From:]
To: <sip:sip:john@jambonz.org>[peer_tag_param]
[last_Call-ID:]
CSeq: 2 ACK
Subject: uac-sip-uri-auth-success
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<!-- Pause briefly -->
<pause milliseconds="2000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:john@172.39.0.24 SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
[last_From:]
[last_To:]
[last_Call-ID:]
CSeq: 3 BYE
Max-Forwards: 70
Subject: uac-sip-uri-auth-success
Content-Length: 0
]]>
</send>
<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>

155
test/scenarios/uas-auth.xml Normal file
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@@ -0,0 +1,155 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- This program is free software; you can redistribute it and/or -->
<!-- modify it under the terms of the GNU General Public License as -->
<!-- published by the Free Software Foundation; either version 2 of the -->
<!-- License, or (at your option) any later version. -->
<!-- -->
<!-- This program is distributed in the hope that it will be useful, -->
<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of -->
<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the -->
<!-- GNU General Public License for more details. -->
<!-- -->
<!-- You should have received a copy of the GNU General Public License -->
<!-- along with this program; if not, write to the -->
<!-- Free Software Foundation, Inc., -->
<!-- 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA -->
<!-- -->
<!-- Sipp default 'uas' scenario. -->
<!-- -->
<scenario name="Basic UAS responder">
<!-- 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 request="INVITE" crlf="true">
</recv>
<send retrans="500">
<![CDATA[
SIP/2.0 401 Unauthorized
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
WWW-Authenticate: Digest realm="jambonz.org", nonce="4cdbb733645816512687270b83d2ae5d11e4d9d8"
Content-Length: 0
]]>
</send>
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<recv request="INVITE" crlf="true">
<action>
<verifyauth assign_to="authvalid" username="foo" password="bar" />
</action>
</recv>
<nop hide="true" test="authvalid" next="goodauth" />
<send retrans="500">
<![CDATA[
SIP/2.0 403 Forbidden
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>;expires=60
Content-Type: application/sdp
Content-Length: 0
]]>
</send>
<recv request="ACK" next="endit"
rtd="true"
crlf="true">
</recv>
<label id="goodauth"/>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>
</send>
<recv request="ACK"
optional="true"
rtd="true"
crlf="true">
</recv>
<recv request="BYE">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<label id="endit"/>
<!-- Keep the call open for a while in case the 200 is lost to be -->
<!-- able to retransmit it if we receive the BYE again. -->
<timewait milliseconds="4000"/>
<!-- 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>

4
test/sip-tests.js
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@@ -54,6 +54,10 @@ test('sbc-outbound tests', async(t) => {
await sippUac('uac-pcap-carrier-success-reinvite.xml');
t.pass('successfully handled reinvite during lcr outdial');
/* invite to sipUri that challenges */
await sippUac('uac-sip-uri-auth-success.xml');
t.pass('successfully connected to sip uri that requires auth');
srf.disconnect();
} catch (err) {
console.error(err);