Hi all,
I'm new to sipp and I'm using sipp 3.4.1 and asterisk 1.8. I met a
problem that I can't find/capture rtp streams on my server side.
The scenario setup is:
uac(IP:10.0.0.11)---------> asterisk(10.0.0.12) ------------------->
uas(10.0.0.13)
And the whole process (which I hope) is like this:
uac
invite ----------------->
100 <----------------
401 <-----------------
ACK ------------------> uac first go through auth process with
asterisk
INVITE ---------------> uas
100 <-------------------
180 <------------------- uas
200 <----------------- uas
ACK -------------------> uas
pause and play pcap for 23 secs
BYE -------------------> uas
200 <------------------ uas
Uac sends invite and play a pcap file, which I wish to capture on uas
side.
Here is the problem, I cannot find/capture any rtp stream on uas side
with wireshark. But I can hear the voice when using a softphone (Blink).
I guess something is wrong with my uas.xml. Please help me!
uas.xml:
<?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>
<!-- 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=[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[local_ip]:[local_port];transport=[transport]>
Content-Length: 0
]]>
</send>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[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>
<!--<pause milliseconds="23313"/>-->
<recv request="ACK"
rtd="true"
crlf="true">
</recv>
<pause milliseconds="23313"/>
<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. -->
<pause milliseconds="1500"/>
<!-- 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>
My uac.xml:
<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">
<!-- You will need to compile SIPp with OpenSSL support 'make ossl' to
use this call scenario -->
<!-- Execute this script with SIPp using the following command assuming
your UAS is 10.0.0.10 -->
<!-- Replace 10.0.0.10 with your SIP proxy’s address. The command will
generate 10 calls (-r) per 10000 -->
<!-- milliseconds (-rp), max 100 concurrent calls (-l) and make a max of
100000 calls (-m) -->
<!-- ./sipp 10.0.0.10 -sf invite-auth-sdp-nomedia.xml -inf
user-accounts.csv -m 100000 -l 100 -r 10 -rp 10000 -->
<scenario name="UAC INVITE with Auth and SDP">
<send retrans="500" start_txn="invite">
<![CDATA[
INVITE sip:[field3]@[field1]:[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:[field3]@[field1]:[remote_port]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
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 response="100" optional="true" rrs="true" response_txn="invite">
</recv>
<recv response="401" auth="true" rrs="true" response_txn="invite">
</recv>
<send ack_txn="invite">
<![CDATA[
ACK sip:[field3]@[field1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
From: sipp <sip:sipp@
[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
[routes]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Content-Length: 0
]]>
</send>
<send retrans="500" start_txn="invite2">
<![CDATA[
INVITE sip:[field3]@[field1]:[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:[field3]@[field1]:[remote_port]>
[routes]
Call-ID: [call_id]
CSeq: [cseq] INVITE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
[field2]
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 response="100" optional="true" response_txn="invite2">
</recv>
<recv response="180" optional="true" response_txn="invite2">
</recv>
<!--<recv response="183" optional="true" response_txn="invite2">
</recv>-->
<!-- Grab the 200 OK's Contact Header's URI for use in the ACK's Request
URI -->
<recv response="200" rtd="true" rrs="true" response_txn="invite2">
<!--<action>
<ereg regexp= "sip:[^;>]+" search_in="hdr" header="Contact:"
assign_to="1" />
</action>-->
</recv>
<!-- decrement the ACK's branch by 3 to match previous INVITE's branch -->
<send ack_txn="invite2">
<![CDATA[
ACK sip:[field3]@[field1] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
From: sipp <sip:sipp@
[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
[last_To]
[routes]
Call-ID: [call_id]
CSeq: [cseq] ACK
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="ref.pcap"/>
</action>
</nop>
<!-- Pause 90 seconds, which is approximately the duration of the -->
<!-- PCAP file -->
<pause milliseconds="23313"/>
<send retrans="500">
<![CDATA[
BYE sip:[field3]@[field1]:[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]
[last_To]
[routes]
Call-ID: [call_id]
CSeq: [cseq] BYE
Contact: <sip:sipp@[local_ip]:[local_port]>
Max-Forwards: 70
Subject: Performance Test
User-Agent: SIPp Tester UAC
[field2]
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
<pause milliseconds="1500">
<!-- 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>
Before start the test, both sides first register to server (expire after
1200 secs) and then using the above two xml file to start calling. I can
hear the voice using Blink but I cannot find/capture rtp with wireshark on
uas side.
I guess somethings wrong with my uas.xml.
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