implement a wav play into pcap

CHANGELOG.md

@@ -11,8 +11,8 @@
   * Feature: Permit `To` domain to be different from the destination. This permits testing multi-tenant systems more easily.
 
 # [0.6.0](https://github.com/mojolingo/sippy_cup/compare/v0.5.0...v0.6.0)
-  * Change: Call limits (`number_of_calls`, `concurrent_max` and `calls_per_second`) no longer have default values for simplicity of UAS scenarios. The value of `to_user` now defaults to the SIPp default of `s`.
-  * Feature: Support for setting rate scaling independently of reporting frequency via the new `calls_per_second_interval` option. See also https://github.com/SIPp/sipp/pull/107 and https://github.com/SIPp/sipp/pull/126.
+  * Change: Call limits (`number_of_calls`, `concurrent_max` and `call_rate`) no longer have default values for simplicity of UAS scenarios. The value of `to_user` now defaults to the SIPp default of `s`.
+  * Feature: Support for setting rate scaling independently of reporting frequency via the new `call_rate_interval` option. See also https://github.com/SIPp/sipp/pull/107 and https://github.com/SIPp/sipp/pull/126.
 
 # [0.5.0](https://github.com/mojolingo/sippy_cup/compare/v0.4.1...v0.5.0)
 SYNTAX CHANGES!

README.markdown

@@ -6,6 +6,15 @@
 
 # Sippy Cup
 
+## NOTES for this fork
+
+* New instruction is added to insert wav file into pcap
+  -  `- play_audio "<wav file path>"`
+  - need spandsp library for encoidng a-law and u-law
+* Debug the DTMF packet generation (end of event) 
+  - reduce the duration to 200 milliseconds
+  - change obsolete rfc2833 to rfc4733
+
 ## Overview
 
 ### The Problem
@@ -78,7 +87,7 @@ Using `bundle` will then install the gem dependencies and allow you to run `sipp
 source: 192.0.2.15
 destination: 192.0.2.200
 max_concurrent: 10
-calls_per_second: 5
+call_rate: 5
 number_of_calls: 20
 steps:
   - invite
@@ -223,7 +232,7 @@ Each parameter has an impact on the test, and may either be changed once the XML
   <dd>By default, SIPp assigns RTP ports dynamically. However, if there is a need for a static RTP port (say, for data collection purposes), it can be done by supplying a port number here. Default: SIPp's default of 6000</dd>
 
   <dt>dtmf_mode</dt>
-  <dd>Specify the mechanism by which DTMF is signaled. Valid options are `rfc2833` for within the RTP media, or `info` for SIP INFO. Default: rfc2833</dd>
+  <dd>Specify the mechanism by which DTMF is signaled. Valid options are `rfc4733` for within the RTP media, or `info` for SIP INFO. Default: rfc4733</dd>
 
   <dt>scenario_variables</dt>
   <dd>If you're using sippy_cup to run a SIPp XML file, there may be CSV fields in the scenario ([field0], [field1], etc.). Specify a path to a CSV file containing the required information using this option. (File is semicolon delimeted, information can be found [here](http://sipp.sourceforge.net/doc/reference.html#inffile).) Default: unused</dd>
@@ -235,19 +244,19 @@ Each parameter has an impact on the test, and may either be changed once the XML
   <dd>The total number of calls permitted for the entire test. When this limit is reached, the test is over. Defaults to nil.</dd>
 
   <dt>concurrent_max</dt>
-  <dd>The maximum number of calls permitted to be active at any given time. When this limit is reached, SIPp will slow down or stop sending new calls until there it falls below the limit. Defaults to SIPp's default: (3 * call_duration (seconds) * calls_per_second)</dd>
+  <dd>The maximum number of calls permitted to be active at any given time. When this limit is reached, SIPp will slow down or stop sending new calls until there it falls below the limit. Defaults to SIPp's default: (3 * call_duration (seconds) * call_rate)</dd>
 
-  <dt>calls_per_second</dt>
+  <dt>call_rate</dt>
   <dd>The rate at which new calls should be created. Note that SIPp will automatically adjust this downward to stay at or beneath the maximum number of concurrent calls (`concurrent_max`). Defaults to SIP's default of 10</dt>
 
-  <dt>calls_per_second_incr</dt>
-  <dd>When used with `calls_per_second_max`, tells SIPp the amount by which `calls_per_second` should be incremented. CPS rate is adjusted each `calls_per_second_interval`. Default: 1.</dd>
+  <dt>call_rate_incr</dt>
+  <dd>When used with `call_rate_max`, tells SIPp the amount by which `call_rate` should be incremented. CPS rate is adjusted each `call_rate_interval`. Default: 1.</dd>
 
-  <dt>calls_per_second_interval</dt>
-  <dd>When used with `calls_per_second_max`, tells SIPp the time interval (in seconds) by which calls-per-second should be incremented. Default: Unset; SIPp's default (60s). NOTE: Requires a development build of SIPp; see https://github.com/SIPp/sipp/pull/107</dd>
+  <dt>call_rate_interval</dt>
+  <dd>When used with `call_rate_max`, tells SIPp the time interval (in seconds) by which calls-per-second should be incremented. Default: Unset; SIPp's default (60s). NOTE: Requires a development build of SIPp; see https://github.com/SIPp/sipp/pull/107</dd>
 
-  <dt>calls_per_second_max</dt>
-  <dd>The maximum rate of calls-per-second. Default: unused (`calls_per_second` will not change)</dd>
+  <dt>call_rate_max</dt>
+  <dd>The maximum rate of calls-per-second. Default: unused (`call_rate` will not change)</dd>
 
   <dt>advertise_address</dt>
   <dd>The IP address to advertise in SIP and SDP if different from the bind IP. Default: `source` IP address</dd>

examples/navigate_ivr.yml

@@ -1,7 +1,7 @@
 source: 192.0.2.15
 destination: 192.0.2.200
 max_concurrent: 10
-calls_per_second: 5
+call_rate: 5
 number_of_calls: 20
 steps:
   - invite

examples/simple_call.yml

@@ -1,7 +1,7 @@
 source: 192.0.2.15
 destination: 192.0.2.200
 max_concurrent: 10
-calls_per_second: 5
+call_rate: 5
 number_of_calls: 20
 steps:
   - invite

examples/wait_for_call.yml

@@ -2,7 +2,7 @@
 source: 10.3.18.108
 destination: 10.3.18.134
 max_concurrent: 1
-calls_per_second: 1
+call_rate: 1
 number_of_calls: 1
 steps:
   - wait_for_call

lib/sippy_cup/g711.rb

@@ -0,0 +1,36 @@
+# ffi interface to freeswitch's g711
+require 'ffi'
+
+module SippyCup
+  module G711
+    extend FFI::Library
+    ffi_lib 'spandsp'
+
+    enum :EncodeState, [
+      :G711_ALAW, 0,
+      :G711_ULAW, 1,
+    ]
+
+    class G711State < FFI::Struct
+      layout :mode,  :EncodeState
+    end
+
+    attach_function :g711_encode, [ G711State, :pointer, :pointer, :int ], :int
+
+    def encode(samples)
+      state = G711State.new
+      state[:mode] = :G711_ULAW # u-law only
+
+      iptr = FFI::MemoryPointer.new(:int16, samples.size)
+      optr = FFI::MemoryPointer.new(:uint8, samples.size)
+
+	  #puts samples.join(' ')
+      iptr.write_array_of_type(:int16, :write_int16, samples)
+      g711_encode(state, optr, iptr, samples.size)
+      output = optr.read_array_of_type(:uint8, :read_uint8, samples.size)
+	  #puts output.join(' ')
+      output
+    end
+    module_function :encode
+  end
+end

lib/sippy_cup/media.rb

@@ -1,11 +1,14 @@
 # encoding: utf-8
 require 'ipaddr'
+require 'wavefile'
+require 'ffi'
 require 'sippy_cup/media/pcmu_payload'
 require 'sippy_cup/media/dtmf_payload'
+require 'sippy_cup/g711'
 
 module SippyCup
   class Media
-    VALID_STEPS = %w{silence dtmf}.freeze
+    VALID_STEPS = %w{silence dtmf play}.freeze
     USEC = 1_000_000
     MSEC = 1_000
     attr_accessor :sequence
@@ -48,13 +51,11 @@ def compile!
           (value.to_i / @generator::PTIME).times do
             packet = new_packet
             rtp_frame = @generator.new
-
             # The first RTP audio packet should have the marker bit set
             if first_audio
               rtp_frame.rtp_marker = 1
               first_audio = false
             end
-
             rtp_frame.rtp_timestamp = timestamp += rtp_frame.timestamp_interval
             elapsed += rtp_frame.ptime
             rtp_frame.rtp_sequence_num = sequence_number += 1
@@ -65,24 +66,52 @@ def compile!
           end
         when 'dtmf'
           # value is the DTMF digit to send
-          # append that RFC2833 digit
-          # Assume 0.25 second duration for now
-          count = 250 / DTMFPayload::PTIME
+          # append that RFC4733 digit
+          # Assume 0.2 second duration for now
+          count = 200 / DTMFPayload::PTIME
           count.times do |i|
             packet = new_packet
             dtmf_frame = DTMFPayload.new value
-            dtmf_frame.rtp_marker = 1 if i == 0
-            dtmf_frame.rtp_timestamp = timestamp # Is this correct? This is what Blink does...
-            #dtmf_frame.rtp_timestamp = timestamp += dtmf_frame.timestamp_interval
+            # The first RTP audio packet should have the marker bit set
+            if first_audio
+              rtp_frame.rtp_marker = 1
+              first_audio = false
+            end
+            dtmf_frame.rtp_timestamp = timestamp += dtmf_frame.timestamp_interval
+            elapsed += dtmf_frame.ptime
             dtmf_frame.rtp_sequence_num = sequence_number += 1
             dtmf_frame.rtp_ssrc_id = ssrc_id
-            dtmf_frame.end_of_event = (count == i) # Last packet?
+            dtmf_frame.end_of_event = (i == count-1) # Last packet
             packet.headers.last.body = dtmf_frame.to_bytes
             packet.recalc
             @pcap_file.body << get_pcap_packet(packet, next_ts(start_time, elapsed))
           end
           # Now bump up the timestamp to cover the gap
           timestamp += count * DTMFPayload::TIMESTAMP_INTERVAL
+        when 'play'
+          # value is wav file path
+          wav = WaveFile::Reader.new(value, WaveFile::Format.new(:mono, :pcm_16, 8000))
+          duration = wav.total_sample_frames * 1000 / wav.native_format.sample_rate # in milliseconds
+          (duration / @generator::PTIME).times do |i|
+            packet = new_packet
+            rtp_frame = @generator.new
+            # The first RTP audio packet should have the marker bit set
+            if first_audio
+              rtp_frame.rtp_marker = 1
+              first_audio = false
+            end
+            rtp_frame.rtp_timestamp = timestamp += rtp_frame.timestamp_interval
+            elapsed += rtp_frame.ptime
+            rtp_frame.rtp_sequence_num = sequence_number += 1
+            rtp_frame.rtp_ssrc_id = ssrc_id
+            len = wav.native_format.sample_rate * rtp_frame.ptime / 1000
+            lin_data = wav.read(len).samples
+			enc_data = G711::encode(lin_data)
+            packet.headers.last.body = rtp_frame.header.to_s << enc_data.flatten.pack('c*')
+            packet.recalc
+            @pcap_file.body << get_pcap_packet(packet, next_ts(start_time, elapsed))
+          end
+          wav.close
         else
         end
       end

lib/sippy_cup/runner.rb

@@ -105,17 +105,19 @@ def command_options
       max_concurrent = @scenario_options[:concurrent_max] || @scenario_options[:max_concurrent]
       options[:l] = max_concurrent if max_concurrent
       options[:m] = @scenario_options[:number_of_calls] if @scenario_options[:number_of_calls]
-      options[:r] = @scenario_options[:calls_per_second] if @scenario_options[:calls_per_second]
+      options[:r] = @scenario_options[:call_rate] if @scenario_options[:call_rate]
+      options[:rp] = @scenario_options[:call_period] if @scenario_options[:call_period]
       options[:s] = @scenario_options[:to].to_s.split('@').first if @scenario_options[:to]
 
       options[:i] = @scenario_options[:source] if @scenario_options[:source]
       options[:mp] = @scenario_options[:media_port] if @scenario_options[:media_port]
+      options[:trace_logs] = nil
 
-      if @scenario_options[:calls_per_second_max]
+      if @scenario_options[:call_rate_max]
         options[:no_rate_quit] = nil
-        options[:rate_max] = @scenario_options[:calls_per_second_max]
-        options[:rate_increase] = @scenario_options[:calls_per_second_incr] || 1
-        options[:rate_interval] = @scenario_options[:calls_per_second_interval] if @scenario_options[:calls_per_second_interval]
+        options[:rate_max] = @scenario_options[:call_rate_max]
+        options[:rate_increase] = @scenario_options[:call_rate_incr] || 1
+        options[:rate_interval] = @scenario_options[:call_rate_interval] if @scenario_options[:call_rate_interval]
       end
 
       if @scenario_options[:stats_file]

lib/sippy_cup/scenario.rb

@@ -88,11 +88,11 @@ def self.from_manifest(manifest, options = {})
     # @option options [Integer] :media_port The RTCP (media) port to bind to locally.
     # @option options [String, Numeric] :max_concurrent The maximum number of concurrent calls to execute.
     # @option options [String, Numeric] :number_of_calls The maximum number of calls to execute in the test run.
-    # @option options [String, Numeric] :calls_per_second The rate at which to initiate calls.
+    # @option options [String, Numeric] :call_rate The rate at which to initiate calls.
     # @option options [String] :stats_file The path at which to dump statistics.
     # @option options [String, Numeric] :stats_interval The interval (in seconds) at which to dump statistics (defaults to 1s).
     # @option options [String] :transport_mode The transport mode over which to direct SIP traffic.
-    # @option options [String] :dtmf_mode The output DTMF mode, either rfc2833 (default) or info.
+    # @option options [String] :dtmf_mode The output DTMF mode, either rfc4733 (default) or info.
     # @option options [String] :scenario_variables A path to a CSV file of variables to be interpolated with the scenario at runtime.
     # @option options [Hash] :options A collection of options to pass through to SIPp, as key-value pairs. In cases of value-less options (eg -trace_err), specify a nil value.
     # @option options [Array<String>] :steps A collection of steps
@@ -175,7 +175,7 @@ def invite(opts = {})
 s=-
 c=IN IP[media_ip_type] [media_ip]
 t=0 0
-m=audio [media_port] RTP/AVP 0 101
+m=audio [auto_media_port] RTP/AVP 0 101
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15
@@ -472,6 +472,16 @@ def sleep(seconds)
       @media << "silence:#{milliseconds}" if @media
     end
 
+    #
+    # add audio to pcap from wav file
+    #
+    # @param [String] path of wav file
+    #
+    def play_audio(wav_file)
+      raise "Media not started" unless @media
+      @media << "play:#{wav_file}"
+    end
+
     #
     # Send DTMF digits
     #
@@ -485,12 +495,12 @@ def sleep(seconds)
     #
     def send_digits(digits)
       raise "Media not started" unless @media
-      delay = (0.250 * MSEC).to_i # FIXME: Need to pass this down to the media layer
+      delay = (0.2 * MSEC).to_i # FIXME: Need to pass this down to the media layer
       digits.split('').each do |digit|
         raise ArgumentError, "Invalid DTMF digit requested: #{digit}" unless VALID_DTMF.include? digit
 
         case @dtmf_mode
-        when :rfc2833
+        when :rfc4733
           @media << "dtmf:#{digit}"
           @media << "silence:#{delay}"
         when :info
@@ -518,7 +528,7 @@ def send_digits(digits)
         end
       end
 
-      if @dtmf_mode == :rfc2833
+      if @dtmf_mode == :rfc4733
         pause delay * 2 * digits.size
       end
     end
@@ -773,9 +783,9 @@ def scenario_node
     def parse_args(args)
       if args[:dtmf_mode]
         @dtmf_mode = args[:dtmf_mode].to_sym
-        raise ArgumentError, "dtmf_mode must be rfc2833 or info" unless [:rfc2833, :info].include?(@dtmf_mode)
+        raise ArgumentError, "dtmf_mode must be rfc4733 or info" unless [:rfc4733, :info].include?(@dtmf_mode)
       else
-        @dtmf_mode = :rfc2833
+        @dtmf_mode = :rfc4733
       end
 
       @from_user = args[:from_user] || "sipp"

sippy_cup.gemspec

@@ -19,9 +19,11 @@ Gem::Specification.new do |s|
   s.require_paths = ["lib"]
 
   s.add_runtime_dependency 'packetfu', ["= 1.1.11"] # 1.1.12 introduces a breaking change, removing PacketFu::UDPPacket
-  s.add_runtime_dependency 'nokogiri', ["~> 1.6.0"]
+  s.add_runtime_dependency 'nokogiri', ["~> 1.7.0"]
   s.add_runtime_dependency 'activesupport', [">= 3.0"]
   s.add_runtime_dependency 'psych', ["~> 2.0.1"] unless RUBY_PLATFORM == 'java'
+  s.add_runtime_dependency 'wavefile', [">= 0.8.0"]
+  s.add_runtime_dependency 'ffi'
 
   s.add_development_dependency 'guard-rspec'
   s.add_development_dependency 'rspec', ["~> 3.4"]

spec/sippy_cup/fixtures/test.yml

@@ -3,7 +3,7 @@ name: My test scenario
 source: 192.0.2.15
 destination: 192.0.2.200
 max_concurrent: 10
-calls_per_second: 5
+call_rate: 5
 number_of_calls: 20
 steps:
   - invite