Remove lots of unnecessary RTPstream code and use [media_port] instead.

CHANGES.md

@@ -1,3 +1,17 @@
+Changes in 3.6.0
+================
+
+* Removed unused RTPStream code concerning video streams. Also
+  consolidated the rtpstream audio port usage to reuse the global
+  `[media_port]` instead of the `[rtpstream_audio_port]`.
+  Also the `-min_rtp_port` and `-max_rtp_port` options have been
+  removed. Advantages: cleaner code, fewer scenario variables.
+  Drawbacks: possible ICMP port unreachable messages for RCTP and video.
+  Also, no easy way to discern different streams if you want to bombard
+  a single UAS with multiple RTP streams. (Issue #192, reported by
+  @atsakiridis.)
+
+
 Bugs fixed in 3.5.1
 ===================
 

include/sipp.hpp

@@ -255,8 +255,6 @@ extern int                tcp_readsize            _DEFVAL(65535);
 extern int                hasMedia                _DEFVAL(0);
 #endif
 #ifdef RTP_STREAM
-extern int                min_rtp_port            _DEFVAL(DEFAULT_MIN_RTP_PORT);
-extern int                max_rtp_port            _DEFVAL(DEFAULT_MAX_RTP_PORT);
 extern int                rtp_default_payload     _DEFVAL(DEFAULT_RTP_PAYLOAD);
 extern int                rtp_tasks_per_thread    _DEFVAL(DEFAULT_RTP_THREADTASKS);
 extern int                rtp_buffsize            _DEFVAL(65535);

regress/github-#0196/run

@@ -0,0 +1,32 @@
+#!/bin/sh
+# This regression test is a part of SIPp.
+# Author: Walter Doekes, OSSO B.V., 2016
+#
+# Similar to github-#0192, it checks whether the rtpstream source port
+# is correct, but this time uses the [media_port] instead of a special
+# [rtpstream_audio_port].
+#
+. "`dirname "$0"`/../functions"; init
+
+uac_media_port=5072  # client port
+uas_media_port=5071  # server port
+
+udplisten $uas_media_port >udplisten.log &
+udplisten_job=$!
+trap "kill -9 $udplisten_job 2>/dev/null" EXIT
+
+sippbg -sf uas.xml -p 5070 -key custom_media_port $uas_media_port
+sippfg -m 1 -sf uac.xml 127.0.0.1:5070 -mp $uac_media_port \
+    -timeout 5 -timeout_error >/dev/null 2>&1
+status=$?
+
+test $status -ne 0 && fail "SIPp UAC job failed"
+
+port=`sed -e '/^Connection from/!d;s/.*://' udplisten.log`
+if test "$port" = "$uac_media_port"; then
+    ok
+elif test -n "$port"; then
+    fail "got RTP from wrong source port $port"
+else
+    fail "got no RTP at all"
+fi

regress/github-#0196/uac.xml

@@ -0,0 +1,80 @@
+<?xml version="1.0" encoding="UTF-8" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="oinv-i200-ibye.xml -- wjd 2014">
+
+  <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 some audio to avoid rtptimeout timers kicking in. -->
+  <nop>
+    <action>
+      <exec rtp_stream="beep_1sec_50x160b.alaw,-1,8"/>
+    </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-#0196/uas.xml

@@ -0,0 +1,78 @@
+<?xml version="1.0" encoding="UTF-8" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+<scenario name="iinv-o200-obye.xml -- wjd 2014">
+
+  <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 [custom_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="_"/>
+
+  <!-- Wait for the other side to send some RTP -->
+  <pause milliseconds="1000"/>
+
+  <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

@@ -2182,26 +2182,10 @@ char* call::createSendingMessage(SendingMessage *src, int P_index, char *msg_buf
             break;
         }
 #ifdef RTP_STREAM
-      case E_Message_RTPStream_Audio_Port:
-        {
-          int temp_audio_port= rtpstream_get_audioport (&rtpstream_callinfo);
-          if (!temp_audio_port) {
-            /* Make this a warning instead? */
-            ERROR("cannot assign a free audio port to this call - using 0 for [rtpstream_audio_port]");
-          }
-          dest += snprintf(dest, left, "%d",temp_audio_port);
-        }
-        break;
-      case E_Message_RTPStream_Video_Port:
-        {
-          int temp_video_port= rtpstream_get_videoport (&rtpstream_callinfo);
-          if (!temp_video_port) {
-            /* Make this a warning instead? */
-            ERROR("cannot assign a free video port to this call - using 0 for [rtpstream_video_port]");
-          }
-          dest += snprintf(dest, left, "%d",temp_video_port);
-        }
-        break;
+        case E_Message_RTPStream_Audio_Port: /* DEPRECATED */
+        case E_Message_RTPStream_Video_Port: /* DEPRECATED */
+            dest += sprintf(dest, "%u", media_port);
+            break;
 #endif
         case E_Message_Media_IP_Type:
             dest += snprintf(dest, left, "%s", (media_ip_is_ipv6 ? "6" : "4"));

src/message.cpp

@@ -62,8 +62,9 @@ struct KeywordMap SimpleKeywords[] = {
     {"auto_media_port", E_Message_Auto_Media_Port },
 #endif
 #ifdef RTP_STREAM
-  {"rtpstream_audio_port", E_Message_RTPStream_Audio_Port },
-  {"rtpstream_video_port", E_Message_RTPStream_Video_Port },
+    /* Legacy since 3.6-dev. Actually uses media_port. */
+    {"rtpstream_audio_port", E_Message_RTPStream_Audio_Port },
+    {"rtpstream_video_port", E_Message_RTPStream_Video_Port },
 #endif
     {"media_port", E_Message_Media_Port },
     {"media_ip_type", E_Message_Media_IP_Type },