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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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10
test/docker-compose-testbed.yaml
View File

@@ -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
View File

@@ -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);