Hey guys!
I'm pretty new to sipp and think it is awesome so far from some of the
things I've been able to do.
I had one problem recently though and after working on it for days I
haven't been able to resolve it on my own.
So I had a Polycom phone that I disconnected and have been emulating with
sipp.
I have two scenarios: uac_register.xml and uac_offer.xml
When I run my scenarios in succession I get the desired phone ringing and
audio works in both directions (I send a g711 file after the connection is
made).
The problem is that I get spammed with 401 Unauthorised messages from the
SBC even though it lets me make the call. It is as if my ACK's are simply
not sent right.
I have the exact same problem with my second ACK which tries to acknowledge
the OK sent to me when the call connects. I simply keep getting spammed
with these OK messages.
If someone could point me in the right direction to fix this that would be
awesome!
The scenario files have been attached. I have wireshark traces too if
anyone is interested
I run my files like this:
./sipp -sf uac_register.xml 192.168.175.132:5060 -r 1 -rp 1 -m 1 -i
10.35.1.165:5061 -recv_timeout 5000
./sipp -sf uac_offer.xml 192.168.175.132:5060 -l 1 -d 5000 -mi 10.35.1.165
-mp 15676 -r 1 -rp 4000 -m 1 -i 10.35.1.165:5061 -s 884414035 -p 5061
And I use sipp 3.4.1 if that helps
Thanks heaps!
Alex
<?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 'uac' scenario. -->
<!-- -->
<scenario name="uac_offer.xml">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send retrans="500" start_txn="invite">
<![CDATA[
INVITE sip:[service]@model.ipvs.net;user=phone SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "884414043 884414043" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@model.ipvs.net;user=phone>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: <sip:p0884414043@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
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
a=media-release:hngl5rp9pujvivh05pvigv2hba8ejor025736svbbeur624c2820
a=media-release-con-addr:d2v53161vl4hj0u6h7210000o0
m=audio [media_port] RTP/AVP 18 0 4 8 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:40
a=ptime:20
a=loopback:rtp-media-loopback
a=loopback-source:18 0 4 8
m=audio 18400 RTP/AVP 113
a=loopback:rtp-start-loopback
]]>
</send>
<recv response="100"
optional="true" response_txn="invite" rrs="true">
</recv>
<recv response="180" optional="true" rrs="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="401" rtd="true" auth="true" rrs="true" response_txn="invite">
<action><ereg regexp="tag=.*" search_in="hdr" header="To:" check_it="false" assign_to="1"/>
<log message="The value of routes is [next_url]"/>
</action>
</recv>
<!-- Packet lost can be simulated in any send/recv message by -->
<!-- by adding the 'lost 10"'. Value can be [1-100] percent. -->
<send ack_txn="invite">
<![CDATA[
ACK sip:[service]@model.ipvs.net;user=phone SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];rport;branch=[branch-4]
From: "884414043 884414043" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@model.ipvs.net;user=phone>[$1]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: <sip:p0884414043@[local_ip]:[local_port]>
Route:<sip:[local_ip]:5060;fid=server_1;lr>
Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER
User-Agent:PolycomSoundPointIP-SPIP_650-UA/4.0.4.2906_0004f23478ed
Accept-Language:en
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<!-- <pause milliseconds="30000"/>-->
<send retrans="500">
<![CDATA[
INVITE sip:[service]@model.ipvs.net;user=phone SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "884414043 884414043" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@model.ipvs.net;user=phone>
Call-ID: [call_id]
CSeq: 2 INVITE
Contact: <sip:p0884414043@[local_ip]:[local_port]>
[authentication username=884414043 password=884414043]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]
v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
t=0 0
c=IN IP[media_ip_type] [media_ip]
a=media-release:hngl5rp9pujvivh05pvigv2hba8ejor025736svbbeur624c2820
a=media-release-con-addr:d2v53161vl4hj0u6h7210000o0
m=audio [media_port] RTP/AVP 18 0 4 8
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:40
a=ptime:20
a=loopback:rtp-media-loopback
a=loopback-source:18 0 4 8
m=audio 18400 RTP/AVP 113
a=loopback:rtp-start-loopback
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180"
optional="true">
</recv>
<recv response="183"
optional="true">
</recv>
<recv response="200" >
<!--<action><ereg regexp="tag=.*" search_in="hdr" header="To:" check_it="false" assign_to="1"/>
</action>-->
</recv>
<send>
<![CDATA[
ACK sip:[service]@model.ipvs.net;user=phone SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "88441403 88441403" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@model.ipvs.net;user=phone>[$1]
Call-ID: [call_id]
CSeq: 2 ACK
Contact: sip:p0884414043@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<nop>
<action>
<exec play_pcap_audio="pcap/g711a.pcap"/>
</action>
</nop>
<pause/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[service]@model.ipvs.net;user=phone SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "884414043 884414043" <sip:[email protected]>;tag=[call_number]
To: <sip:[service]@model.ipvs.net;user=phone>[$1]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:p0884414043@[local_ip]:[local_port]
[authentication username=884414043 password=884414043]
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>
<?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 'uac' scenario. -->
<!-- -->
<scenario name="uac_register.xml">
<!-- In client mode (sipp placing calls), the Call-ID MUST be -->
<!-- generated by sipp. To do so, use [call_id] keyword. -->
<send >
<![CDATA[
REGISTER sip:model.ipvs.net SIP/2.0
Via:SIP/2.0/UDP [local_ip]:[local_port];branch=z9hG4bK88855544011;rport
To: <sip:[email protected]>
From:"884414043 884414043" <sip:[email protected]>;tag=[call_number]
Call-ID: [call_id]
CSeq: 1 REGISTER
Contact: <sip:p0884414043@[local_ip]:[local_port]>
Content-Length: 0
Max-Forwards: 70
Expires:3600
]]>
</send>
<recv response="200" optional="true">
</recv>
<recv response="401" auth="true">
</recv>
<send>
<![CDATA[
REGISTER sip:model.ipvs.net SIP/2.0
Via:SIP/2.0/UDP [local_ip]:[local_port];branch=z9hG4bK88855544011;rport
To: <sip:[email protected]>
From: "884414043 884414043"<sip:[email protected]>;tag=[call_number]
Contact: <sip:p0884414043@[local_ip]:[local_port]>
[authentication username=884414043 password=884414043]
Call-ID: [call_id]
CSeq: 2 REGISTER
Max-Forwards: 70
Expires:3600
Content-Length: 0
]]>
</send>
<recv response="200" >
</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>
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