DeforaNetworks/siproxd
DeforaNetworks/siproxd
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This commit is contained in:
Thomas Ries 2010-01-07 11:31:17 +00:00
parent 593ec99816
commit 7dda94215d
4 changed files with 293 additions and 2 deletions
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6
src/resolve.c
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@ -33,6 +33,8 @@
#include "log.h"
#define USE_NAPTR 0
static char const ident[]="$Id$";
@ -54,7 +56,7 @@ int resolve_SRV(char *name, char *dname, int dnamelen, int *port) {
return _resolve(name, C_IN, T_SRV, dname, dnamelen, port);
}
#if 0
#if USE_NAPTR
/*
* perform a NAPTR lookup
*
@ -186,7 +188,7 @@ Currently just the first (lowest prio, highest weight) entry is returned.
xptr+=j;
}
}
#if 0
#if USE_NAPTR
} else if( ty == T_NAPTR ) {
DEBUGC(DBCLASS_DNS, "_resolve: A - type NAPTR");
usp = (unsigned short *)xptr;

36
tools/sipp_testing/README Normal file
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@ -0,0 +1,36 @@
Setup:
------
HOSTA ---------- siproxd ------------HOSTB
private net | Internet
siproxd must be setup as transparent proxy (iptables rules to
redirect all outgoing SIP traffic from sipp to HOSTB to local siproxd)
Client on HOSTA
---------------
- limited to 10 simultaneous calls (-> default configured RTP port range)
- server and client are on port 5070
- each call is kept for 1000ms (-d), then terminated
- every 1000ms (-rp) 5 (-r) new calls are spawned
- end test after 100 total calls
./sipp -sf myuac.xml -l 10 -p 5070 -d 1000 \
-m 100 -r 5 -rp 1000 HOSTB:5070
Server on HOSTB
---------------
- port 5070
./sipp -sf myuas.xml -p 5070
Note: siproxd has limited simultaneous calls
- RTP port range
- some compile time constants for RTP array sizing
Those XML scenarios include a REGISTER step before doing the
actual call. This is required for siproxd to know the local UA.

128
tools/sipp_testing/myuac.xml Normal file
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@ -0,0 +1,128 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- -->
<!-- Sipp default 'myuac' scenario with REGISTER -->
<!-- -->
<scenario name="Basic Sipstone UAC">
<!-- 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[
REGISTER sip:[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag07[call_number]
To: sipp <sip:sipp@[local_ip]:[local_port]>
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: <sip:sipp@[local_ip]:[local_port]>
Content-Length: 0
Expires: 300
]]>
</send>
<recv response="200">
</recv>
<send retrans="500">
<!--bloody hack to get the CRLF to the end of the SDP, [len+2] -->
<![CDATA[
INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 151
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=session
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
foo
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="183" 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">
</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:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
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>

125
tools/sipp_testing/myuas.xml Normal file
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@ -0,0 +1,125 @@
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- -->
<!-- Sipp 'myuas' scenario, with REGISTER. -->
<!-- -->
<scenario name="Basic UAS responder">
<recv request="REGISTER">
</recv>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag08[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
Expires: 300
]]>
</send>
<!-- 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>
<!-- The '[last_*]' keyword is replaced automatically by the -->
<!-- specified header if it was present in the last message received -->
<!-- (except if it was a retransmission). If the header was not -->
<!-- present or if no message has been received, the '[last_*]' -->
<!-- keyword is discarded, and all bytes until the end of the line -->
<!-- are also discarded. -->
<!-- -->
<!-- If the specified header was present several times in the -->
<!-- message, all occurences are concatenated (CRLF seperated) -->
<!-- to be used in place of the '[last_*]' keyword. -->
<send>
<![CDATA[
SIP/2.0 180 Ringing
[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-Length: 0
]]>
</send>
<send retrans="500">
<!--bloody hack to get the CRLF to the end of the SDP, [len+2] -->
<![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: 155
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=session
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
foo
]]>
</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>
<!-- 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>