From 3cdb9eab57dac8497d13e3a9b90cf40505db767f Mon Sep 17 00:00:00 2001
From: Peter Wu <peter@lekensteyn.nl>
Date: Fri, 2 Dec 2016 11:24:23 +0100
Subject: Added SIPp scenario and list of codecs supported by FS

Requires appropriately configured FreeSWITCH server that responds to a
call to sip:test@host by playing a fragment, then hanging up.

SIPp scenario was used to create a bunch of captures, uploaded to
https://wiki.wireshark.org/SampleCaptures#SIP_and_RTP
---
 sipsim/codecs.txt    | 57 ++++++++++++++++++++++++++++++++
 sipsim/uac_media.xml | 92 ++++++++++++++++++++++++++++++++++++++++++++++++++++
 2 files changed, 149 insertions(+)
 create mode 100644 sipsim/codecs.txt
 create mode 100644 sipsim/uac_media.xml

diff --git a/sipsim/codecs.txt b/sipsim/codecs.txt
new file mode 100644
index 0000000..94cac76
--- /dev/null
+++ b/sipsim/codecs.txt
@@ -0,0 +1,57 @@
+
+# Note: codecname[@8000h|16000h|32000h[@XXi]]
+
+# mod_g723_1 is for passthrough
+# G723/8000 4
+
+# (in core)
+#PCMU/8000 0
+#PCMA/8000 8
+# RFC 3555 valid sample sizes for L16: 8000, 11025, 16000, 22050, 24000, 32000, 44100, and 48000
+# L16 RTP clock rate is always 44100. (pt 10 and 11, but dynamic for others)
+#L16/8000/2
+#L16/16000/2
+#L16/11025
+#L16/48000
+
+# mod_com_g728
+# G728/8000 15
+
+# mod_com_g729
+# G729/8000 18
+
+
+# (mostly) dynamic types
+# mod_ilbc
+#iLBC/8000 # note: a=fmtp:97 mode=20 or mode=30
+
+# mod_siren
+#G7221/16000 # note: a=fmtp:121 bitrate=24000
+#G7221/32000 # note: a=fmtp:122 bitrate=48000
+
+# note, additionally also supported: PCMU/PCMA (G711)
+# mod_spandsp
+#GSM/8000 3
+#DVI4/8000 5
+#DVI4/16000 6
+#LPC/8000 7
+# G722/8000 on wire, but actual sample size is 16kHz
+# G722/8000 9
+#G722/16000
+#G726-16/8000
+#G726-24/8000
+#G726-32/8000
+#G726-40/8000
+#AAL2-G726-16/8000
+#AAL2-G726-24/8000
+#AAL2-G726-32/8000
+#AAL2-G726-40/8000
+
+# Maybe FS wants capital OPUS and SPEEX...
+# mod_opus
+#opus/48000/2
+
+# (in core)
+#speex/8000
+#speex/16000
+#speex/32000
diff --git a/sipsim/uac_media.xml b/sipsim/uac_media.xml
new file mode 100644
index 0000000..51b5433
--- /dev/null
+++ b/sipsim/uac_media.xml
@@ -0,0 +1,92 @@
+<?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             -->
+<!--                                                                    -->
+<!--        Based on sipp default 'uac' scenario.                       -->
+<!-- To make one call to test@10.0.2.15 from IP 10.0.2.20 using PCMU codec:
+     ./sipp -sf uac_media.xml 10.0.2.15 -i 10.0.2.20 -bind_local -m 1 -s test -key codec PCMU/8000 -key pt 99
+  -->
+<!--                                                                    -->
+
+<scenario name="Codec test">
+  <send retrans="500">
+    <![CDATA[
+      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: "[codec]" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=- 42 42 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 [pt]
+      a=rtpmap:[pt] [codec]
+      a=recvonly
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <recv response="200" rtd="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: "[codec]" <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
+      To: [service] <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
+      Content-Length: 0
+    ]]>
+  </send>
+
+  <!-- wait for callee to hang up -->
+  <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>
+</scenario>
-- 
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