Hi all,
I have following issue: “\r\n” characters are added in ACK’s SDP part:
***************************
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
******************************
Output:
[cid:image001.png@01CF9A8D.23EB8EA0]
I tried dos2unix, re-created the xml SDP via vi editor. But didn’t get any
success. Used XML are attached.
Any idea? Is this a bug?
SIPp v3.4.1-TLS-SCTP-PCAP-RTPSTREAM built
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<?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:[field3]@[field0] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[field1]" <sip:[field1]@[field2]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field3]@[field2]>
Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
Call-ID: [call_id]
CSeq: [cseq] INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 0
]]>
</send>
<recv response="100"
optional="true">
</recv>
<recv response="180" optional="true">
</recv>
<recv response="181" 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 [remote_ip]:[remote_port] -->
<!-- by adding the 'lost = "10"'. Value can be [1-100] percent. -->
<send retrans="500">
<![CDATA[
ACK sip:[field3]@[field0]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
Max-Forwards: 70
Contact: <sip:[field1]@[local_ip]:[local_port]>;transport=[transport]
From: "[field1]" <sip:[field1]@[field2]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field3]@[field2]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: [cseq] ACK
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>
<!-- This delay can be customized by the -d command-line option -->
<!-- or by adding a 'milliseconds = "value"' option here. -->
<pause milliseconds="2000"/>
<!-- The 'crlf' option inserts a blank line in the statistics report. -->
<send retrans="500">
<![CDATA[
BYE sip:[field3]@[field0]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: "[field1]" <sip:[field1]@[field2]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: <sip:[field3]@[field2]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: [cseq] BYE
Contact: sip:[field1]@[local_ip]:[local_port];transport=[transport]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>
</send>
<recv response="200" crlf="true">
</recv>
</scenario>
<?xml version="1.0" encoding="ISO-8859-1"?>
<!DOCTYPE scenario PUBLIC "-//sipp" "">
<scenario
name="TERMINATOR - ">
<recv request="INVITE"/>
<send>
<![CDATA[
SIP/2.0 100 Trying
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Content-Length: 0
]]>
</send>
<send>
<![CDATA[
SIP/2.0 180 Ringing
[last_Via:]
Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
[last_From:]
[last_To:];tag=[pid]
[last_Call-ID:]
[last_CSeq:]
User-Agent: Asterik
Content-Length: 0
]]>
</send>
<pause milliseconds="500"/>
<send retrans="500">
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]
[last_Call-ID:]
[last_CSeq:]
Contact: <sip:[field1]@[local_ip]:[local_port];transport=[transport]>
Content-Type: application/sdp
Content-Length: [len]
v=0
o=- 6 1 IN IP[local_ip_type] [local_ip]
s=Asterisk
c=IN IP[local_ip_type] [local_ip]
t=0 0
m=audio 50026 RTP/AVP 0 97
a=fmtp:97 0-15
a=rtpmap:97 telephone-event/8000
a=sendrecv
]]>
</send>
<recv request="ACK"/>
<!-- Speech path -->
<recv request="BYE"/>
<send>
<![CDATA[
SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Content-Length: 0
]]>
</send>
</scenario>
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