Added the rtp_echo scenario action

CHANGES.md

@@ -31,7 +31,10 @@ Features added in 3.6.0
     and must be between 50 and 2000.
   - Instead of digits a `[field...]` keyword is also accepted.
   - Make sure you add enough `<pause/>` after `play_dtmf`.
-
+* Added `rtp_echo` action (pull #259 by Snom Technology). Compile with
+  `--with-rtpstream` and use it by adding `<rtp_echo value="0">` to stop
+  the RTP echo enabled via `-rtp_echo`. RTP echo can be restarted via
+  `<rtp_echo value="1">` action. Usage example in `regress/github-#0259/uas.xml`
 
 Bugs fixed in 3.5.x
 ===================

include/actions.hpp

@@ -79,6 +79,7 @@ class CAction
         E_AT_RTP_STREAM_PAUSE,
         E_AT_RTP_STREAM_RESUME,
         E_AT_RTP_STREAM_PLAY,
+        E_AT_RTP_ECHO,
 #endif
         E_AT_NB_ACTION
     };

include/sipp.hpp

@@ -300,6 +300,9 @@ extern char             * auth_uri                _DEFVAL(0);
 extern const char       * call_id_string          _DEFVAL("%u-%p@%s");
 extern char             **generic[100];
 
+#ifdef RTP_STREAM
+extern bool               rtp_echo_state          _DEFVAL(true);
+#endif
 extern bool               callidSlash             _DEFVAL(false);
 
 /* TDM map */

regress/github-#0259/run

@@ -0,0 +1,20 @@
+#!/bin/sh
+# This regression test is a part of SIPp.
+# Author: Pietro Bertera, Snom Technology AG, 2016
+#
+# Start the UAS scenario, UAS stops the rtp_echo 
+# for some seconds
+#
+. "`dirname "$0"`/../functions"; init
+
+if ! sippversion | grep -q RTPSTREAM; then
+    skip "requires RTPSTREAM"
+fi
+
+sippbg -sf uas.xml -p 5070 -rtp_echo
+sippfg -m 1 -sf uac.xml 127.0.0.1:5070 \
+    -timeout 10 -timeout_error >/dev/null 2>&1
+status=$?
+
+test $status -ne 0 && fail "SIPp UAC job failed"
+ok

regress/github-#0259/uac.xml

@@ -0,0 +1,80 @@
+<?xml version="1.0" encoding="UTF-8" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="RTP Echo pause / start test UAC">
+
+  <send retrans="500" start_txn="invite">
+    <![CDATA[
+
+      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:[service]@[remote_ip]:[remote_port]
+      Call-ID: [call_id]
+      CSeq: [cseq] INVITE
+      Contact: sip:[service]@[local_ip]:[local_port]
+      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 8 0
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true" response_txn="invite"/>
+
+  <recv response="180" optional="true" response_txn="invite"/>
+
+  <recv response="183" optional="true" response_txn="invite"/>
+
+  <recv response="200" rrs="true" response_txn="invite"/>
+
+  <send ack_txn="invite">
+    <![CDATA[
+
+      ACK [next_url] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      [routes]
+      From: sip:[service]@[local_ip]:[local_port];tag=[pid]SIPpTag00[call_number]
+      To: sip:[service]@[remote_ip]:[remote_port][peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: [cseq] ACK
+      Contact: sip:[service]@[local_ip]:[local_port]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Send the audio for 8 seconds -->
+  <nop>
+    <action>
+      <exec rtp_stream="beep_1sec_50x160b.alaw,8,8"/>
+    </action>
+  </nop>
+
+  <recv request="BYE"/>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_Record-Route:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <timewait milliseconds="1000"/><!-- retransmit 200's for a while -->
+
+</scenario>

regress/github-#0259/uas.xml

@@ -0,0 +1,93 @@
+<?xml version="1.0" encoding="UTF-8" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="RTP Echo pause / start test UAS">
+
+  <recv request="INVITE" rrs="true"/>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 180 Ringing
+      [last_Via:]
+      [last_Record-Route:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_Record-Route:]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Contact: sip:[service]@[local_ip]:[local_port]
+      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 8 0
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv request="ACK" rrs="true">
+    <action>
+      <ereg regexp=": *(.*)" search_in="hdr" header="To" assign_to="_,us"/><!-- with tag -->
+      <ereg regexp=": *(.*)" search_in="hdr" header="From" assign_to="_,them"/>
+    </action>
+  </recv>
+  <Reference variables="_"/>
+
+  <pause milliseconds="2000"/>
+
+  <nop>
+    <action>
+      <rtp_echo value="0"/> <!-- stop the RTP echo -->
+    </action>
+  </nop>
+
+  <pause milliseconds="2000"/>
+
+  <nop>
+    <action>
+      <rtp_echo value="1"/> <!-- start the RTP echo -->
+    </action>
+  </nop>
+
+  <pause milliseconds="2000"/>
+
+  <send start_txn="bye" retrans="500">
+    <![CDATA[
+
+      BYE [next_url] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      [routes]
+      From: [$us]
+      To: [$them]
+      [last_Call-ID:]
+      CSeq: [cseq] BYE
+      Contact: sip:[service]@[local_ip]:[local_port]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" response_txn="bye"/>
+
+</scenario><!-- vim: set ts=8 sw=2 sts=2 et ai: -->

src/call.cpp

@@ -3691,6 +3691,8 @@ call::T_ActionResult call::executeAction(char * msg, message *curmsg)
 #endif
 
 #ifdef RTP_STREAM
+        } else if (currentAction->getActionType() == CAction::E_AT_RTP_ECHO) {
+            rtp_echo_state = (currentAction->getDoubleValue() != 0);
         } else if (currentAction->getActionType() == CAction::E_AT_RTP_STREAM_PAUSE) {
             rtpstream_pause(&rtpstream_callinfo);
         } else if (currentAction->getActionType() == CAction::E_AT_RTP_STREAM_RESUME) {

src/scenario.cpp

@@ -1647,6 +1647,13 @@ void scenario::parseAction(CActions *actions)
             } else {
                 ERROR("illegal <exec> in the scenario\n");
             }
+        } else if(!strcmp(actionElem, "rtp_echo")) {
+#ifdef RTP_STREAM
+            tmpAction->setActionType(CAction::E_AT_RTP_ECHO);
+            handle_rhs(tmpAction, "rtp_echo");
+#else
+            ERROR("Scenario specifies a rtp_echo action, but this version of SIPp does not have RTP stream support");
+#endif
         } else {
             ERROR("Unknown action: %s", actionElem);
         }

src/sipp.cpp

@@ -603,6 +603,11 @@ static void rtp_echo_thread(void* param)
                     errno);
             return;
         }
+#ifdef RTP_STREAM
+        if (!rtp_echo_state) {
+            continue;
+        }
+#endif
         ns = sendto(*(int*)param, msg, nr, 0,
                     (sockaddr*)&remote_rtp_addr, len);