From b0760f8006153d3db23a2026777ada3a10bf8f9c Mon Sep 17 00:00:00 2001
From: Mark Spencer <markster@digium.com>
Date: Fri, 16 May 2003 02:50:46 +0000
Subject: [PATCH] Make RTP ports configurable

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1026 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 CHANGES                 |  1 +
 asterisk.c              |  2 ++
 configs/rtp.conf.sample |  9 +++++++
 include/asterisk/rtp.h  |  4 +++
 loader.c                |  2 ++
 rtp.c                   | 59 ++++++++++++++++++++++++++++++++++++++---
 6 files changed, 74 insertions(+), 3 deletions(-)
 create mode 100755 configs/rtp.conf.sample

diff --git a/CHANGES b/CHANGES
index 7535071abb..2ca1a3e1af 100755
--- a/CHANGES
+++ b/CHANGES
@@ -1,3 +1,4 @@
+ -- Make RTP ports configurable
  -- Add RDNIS support to SIP and IAX2
  -- Add transfer app (implement in SIP and IAX2)
  -- Make voicemail segmentable by context (app_voicemail2)
diff --git a/asterisk.c b/asterisk.c
index 70cd1a88d4..933ab55947 100755
--- a/asterisk.c
+++ b/asterisk.c
@@ -27,6 +27,7 @@
 #include <asterisk/manager.h>
 #include <asterisk/pbx.h>
 #include <asterisk/enum.h>
+#include <asterisk/rtp.h>
 #include <sys/resource.h>
 #include <fcntl.h>
 #include <stdio.h>
@@ -1317,6 +1318,7 @@ int main(int argc, char *argv[])
 		printf(term_quit());
 		exit(1);
 	}
+	ast_rtp_init();
 	if (ast_image_init()) {
 		printf(term_quit());
 		exit(1);
diff --git a/configs/rtp.conf.sample b/configs/rtp.conf.sample
new file mode 100755
index 0000000000..01a6dea891
--- /dev/null
+++ b/configs/rtp.conf.sample
@@ -0,0 +1,9 @@
+;
+; RTP Configuration
+;
+[general]
+;
+; RTP start and RTP end configure start and end addresses
+;
+rtpstart=10000
+rtpend=20000
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index e0b83bb0f3..c183cb1bdb 100755
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -99,6 +99,10 @@ void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto);
 
 void ast_rtp_stop(struct ast_rtp *rtp);
 
+void ast_rtp_init(void);
+
+void ast_rtp_reload(void);
+
 #if defined(__cplusplus) || defined(c_plusplus)
 }
 #endif
diff --git a/loader.c b/loader.c
index b10a903875..757352f78e 100755
--- a/loader.c
+++ b/loader.c
@@ -24,6 +24,7 @@
 #include <asterisk/term.h>
 #include <asterisk/manager.h>
 #include <asterisk/enum.h>
+#include <asterisk/rtp.h>
 #include <dlfcn.h>
 #include <asterisk/md5.h>
 #include <pthread.h>
@@ -146,6 +147,7 @@ void ast_module_reload(void)
 	/* We'll do the logger and manager the favor of calling its reload here first */
 	reload_manager();
 	ast_enum_reload();
+	ast_rtp_reload();
 	time(&ast_lastreloadtime);
 
 	ast_pthread_mutex_lock(&modlock);
diff --git a/rtp.c b/rtp.c
index 90266f837e..63cb6c415d 100755
--- a/rtp.c
+++ b/rtp.c
@@ -32,6 +32,7 @@
 #include <asterisk/channel.h>
 #include <asterisk/acl.h>
 #include <asterisk/channel_pvt.h>
+#include <asterisk/config.h>
 
 #define TYPE_HIGH	 0x0
 #define TYPE_LOW	 0x1
@@ -41,6 +42,9 @@
 
 static int dtmftimeout = 300;	/* 300 samples */
 
+static int rtpstart = 0;
+static int rtpend = 0;
+
 // The value of each payload format mapping:
 struct rtpPayloadType {
   int isAstFormat; // whether the following code is an AST_FORMAT
@@ -567,6 +571,7 @@ struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
 	struct ast_rtp *rtp;
 	int x;
 	int flags;
+	int startplace;
 	rtp = malloc(sizeof(struct ast_rtp));
 	if (!rtp)
 		return NULL;
@@ -583,11 +588,12 @@ struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
 	}
 	flags = fcntl(rtp->s, F_GETFL);
 	fcntl(rtp->s, F_SETFL, flags | O_NONBLOCK);
+	/* Find us a place */
+	x = (rand() % (rtpend-rtpstart)) + rtpstart;
+	x = x & ~1;
+	startplace = x;
 	for (;;) {
-		/* Find us a place */
-		x = (rand() % (65000-1025)) + 1025;
 		/* Must be an even port number by RTP spec */
-		x = x & ~1;
 		rtp->us.sin_port = htons(x);
 		if (!bind(rtp->s, &rtp->us, sizeof(rtp->us)))
 			break;
@@ -597,6 +603,15 @@ struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
 			free(rtp);
 			return NULL;
 		}
+		x += 2;
+		if (x > rtpend)
+			x = (rtpstart + 1) & ~1;
+		if (x == startplace) {
+			ast_log(LOG_WARNING, "No RTP ports remaining\n");
+			close(rtp->s);
+			free(rtp);
+			return NULL;
+		}
 	}
 	if (io && sched) {
 		/* Operate this one in a callback mode */
@@ -1081,3 +1096,41 @@ int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, st
 	}
 	return -1;
 }
+
+void ast_rtp_reload(void)
+{
+	struct ast_config *cfg;
+	char *s;
+	rtpstart = 5000;
+	rtpend = 31000;
+	cfg = ast_load("rtp.conf");
+	if (cfg) {
+		if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
+			rtpstart = atoi(s);
+			if (rtpstart < 1024)
+				rtpstart = 1024;
+			if (rtpstart > 65535)
+				rtpstart = 65535;
+		}
+		if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
+			rtpend = atoi(s);
+			if (rtpend < 1024)
+				rtpend = 1024;
+			if (rtpend > 65535)
+				rtpend = 65535;
+		}
+		ast_destroy(cfg);
+	}
+	if (rtpstart >= rtpend) {
+		ast_log(LOG_WARNING, "Unreasonable values for RTP start/end\n");
+		rtpstart = 5000;
+		rtpend = 31000;
+	}
+	if (option_verbose > 1)
+		ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
+}
+
+void ast_rtp_init(void)
+{
+	ast_rtp_reload();
+}