Hi,
I use Sipp to send data (play pcap file ) and Asterisk replies messages INFO.
- If I use option "-aa" Sipp answers automatically message INFO by
message 200 OK then Sipp has error "Segmentation fault "
ex : ./sipp -sf sipp_uac.xml -inf database.csv 157.159.16.156:5060 -s
100 -i 157.159.16.91 -trace_err -trace_logs -aa -m $1 -t un
- If I don't use options "-aa", Sipp has a message :
"2010-09-09 14:16:59:951 1284034619.951458: Continuing call on
unexpected message for Call-Id '[email protected]': while pausing
(index 7), received 'INFO sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 157.159.16.156:5060;branch=z9hG4bK26e8be7d;rport
Max-Forwards: 70
From: sut <sip:[email protected]:5060>;tag=as2fdfb442
To: sipp <sip:[email protected]:5061>;tag=1
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INFO
User-Agent: Asterisk PBX 1.6.2.11
Content-Type: application/media_control+xml
Content-Length: 205
<?xml version="1.0" encoding="utf-8" ?>
<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>
'.
sipp: There were more errors, enable -trace_err to log them. "
but Sipp still transfers data video and audio to server.
I don't understand that Asterisk sends message INFO to client Sipp but
with Softphone, Asterisk doesn't send this message INFO.
I have attached my scenario to test Asterisk
Anyone can help me to solve this problem ?
Thank in advance for all helps!
Best regards,
T.Q.Tuan
On Thu, Sep 9, 2010 at 10:10 AM, tran quoc tuan <[email protected]> wrote:
> Hi all,
> I have tested successfully Softphone as client to connect Asterisk as Server.
> I want to test Sipp as client to connect Asterisk as Server, because I
> want to test with more and more users.
> I have tried to create a scenario but it doesn't create a session.
> In Asterisk, it returns message : 488 Not Acceptable here. I don't
> understand this reason, because SDP in message INVITE, I copy from
> message of Softphone.
>
> I have attached my scenario,
> Thank in advance for all helps!
> Best regards,
> T.Q.Tuan
>
<?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="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[
INVITE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[servi...@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:s...@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Dummy User
User-Agent: sipp
Allow: ACK, BYE, CANCEL, INVITE, REFER, OPTIONS, INFO, REGISTER, NOTIFY, UPDATE, SUBSCRIBE, PRACK
Content-Type: application/sdp
Content-Length: [len]
v=0
o=sipp 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 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=sendonly
m=video [auto_media_port+10] RTP/AVP 96
a=fmtp:96 QCIF=1 CIF=1
a=rtpmap:96 H263-1998/90000
a=sendonly
]]>
</send>
<recv response="100" optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<!--
m=audio [auto_media_port] RTP/AVP 0 97 8 18 3 101
a=fmtp:18 annexb=yes
a=fmtp:101 0-11,16
a=rtpmap:0 PCMU/8000
a=rtpmap:97 SPEEX/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendonly
m=video [media_port] RTP/AVP 125 126 115 34
a=fmtp:125 profile-level-id=42e00a; max-br=452; max-mbps=11880
a=fmtp:126 profile-level-id=42e01e; packetization-mode=1; max-br=452; max-mbps=11880
a=fmtp:96 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520
a=fmtp:34 QCIF=1 CIF=1 MaxBR=4520
a=rtpmap:125 H264/90000
a=rtpmap:126 H264/90000
a=rtpmap:96 H263-1998/90000
a=rtpmap:34 H263/90000
a=sendonly
a=fmtp:96 packetization-mode=1;profile-level-id=4d4033;
-->
<!-- 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:[servi...@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:s...@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[servi...@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:s...@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Dummy User
Content-Length: 0
]]>
</send>
<!-- Play a pre-recorded PCAP file (RTP stream) -->
<nop>
<action>
<exec play_pcap_audio="8Khz.pcap"/>
</action>
</nop>
<nop>
<action>
<exec play_pcap_video="paris.pcap"/>
</action>
</nop>
<pause milliseconds="60000" crlf="true" />
<send retrans="500">
<![CDATA[
BYE sip:157.159.16.91:5060 SIP/2.0
[last_Via:]
Max-Forwards: 20
[routes]
From: "[field0]" <sip:[fiel...@[field2]>;tag=[call_number]
To: <sip:[servi...@[remote_ip]:5060>
P-Preferred-Identity: <sip:[fiel...@[field2]>
P-Access-Network-Info: 3GPP-UTRAN-TDD;utran-cell-id-3gpp=C359A3913B20E
Call-ID: [call_id]
CSeq: [cseq] BYE
Content-Length: 0
]]>
</send>
<!-- 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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