[asterisk-commits] trunk r15435 - in /trunk: ./ channels/ configs/
doc/ include/asterisk/
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Mar 27 20:28:54 MST 2006
Author: kpfleming
Date: Mon Mar 27 21:28:52 2006
New Revision: 15435
URL: http://svn.digium.com/view/asterisk?rev=15435&view=rev
Log:
improve IP TOS support for SIP and IAX2 (issue #6355, code from jcollie plus modifications)
Added:
trunk/doc/ip-tos.txt
- copied unchanged from r15430, team/oej/iptos/doc/ip-tos.txt
Modified:
trunk/acl.c
trunk/channels/chan_iax2.c
trunk/channels/chan_sip.c
trunk/channels/iax2-provision.c
trunk/configs/iax.conf.sample
trunk/configs/iaxprov.conf.sample
trunk/configs/sip.conf.sample
trunk/include/asterisk/acl.h
Modified: trunk/acl.c
URL: http://svn.digium.com/view/asterisk/trunk/acl.c?rev=15435&r1=15434&r2=15435&view=diff
==============================================================================
--- trunk/acl.c (original)
+++ trunk/acl.c Mon Mar 27 21:28:52 2006
@@ -1,7 +1,7 @@
/*
* Asterisk -- An open source telephony toolkit.
*
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Copyright (C) 1999 - 2006, Digium, Inc.
*
* Mark Spencer <markster at digium.com>
*
@@ -252,12 +252,55 @@
return 0;
}
-int ast_str2tos(const char *value, int *tos)
+struct dscp_codepoint {
+ char *name;
+ unsigned int space;
+};
+
+/* IANA registered DSCP codepoints */
+
+static const struct dscp_codepoint dscp_pool1[] = {
+ { "CS0", 0x00 },
+ { "CS1", 0x08 },
+ { "CS2", 0x10 },
+ { "CS3", 0x18 },
+ { "CS4", 0x20 },
+ { "CS5", 0x28 },
+ { "CS6", 0x30 },
+ { "CS7", 0x38 },
+ { "AF11", 0x0A },
+ { "AF12", 0x0C },
+ { "AF13", 0x0E },
+ { "AF21", 0x12 },
+ { "AF22", 0x14 },
+ { "AF23", 0x16 },
+ { "AF31", 0x1A },
+ { "AF32", 0x1C },
+ { "AF33", 0x1E },
+ { "AF41", 0x22 },
+ { "AF42", 0x24 },
+ { "AF43", 0x26 },
+ { "EF", 0x2E },
+};
+
+int ast_str2tos(const char *value, unsigned int *tos)
{
int fval;
- if (sscanf(value, "%i", &fval) == 1)
- *tos = fval & 0xff;
- else if (!strcasecmp(value, "lowdelay"))
+ unsigned int x;
+
+ if (sscanf(value, "%i", &fval) == 1) {
+ *tos = fval & 0xFF;
+ return 0;
+ }
+
+ for (x = 0; x < sizeof(dscp_pool1) / sizeof(dscp_pool1[0]); x++) {
+ if (!strcasecmp(value, dscp_pool1[x].name)) {
+ *tos = dscp_pool1[x].space << 2;
+ return 0;
+ }
+ }
+
+ if (!strcasecmp(value, "lowdelay"))
*tos = IPTOS_LOWDELAY;
else if (!strcasecmp(value, "throughput"))
*tos = IPTOS_THROUGHPUT;
@@ -269,7 +312,35 @@
*tos = 0;
else
return -1;
+
+ ast_log(LOG_WARNING, "tos value %s is deprecated. See doc/iptos.txt for more information.", value);
+
return 0;
+}
+
+const char *ast_tos2str(unsigned int tos)
+{
+ unsigned int x;
+
+ switch (tos) {
+ case 0:
+ return "none";
+ case IPTOS_LOWDELAY:
+ return "lowdelay";
+ case IPTOS_THROUGHPUT:
+ return "throughput";
+ case IPTOS_RELIABILITY:
+ return "reliability";
+ case IPTOS_MINCOST:
+ return "mincost";
+ default:
+ for (x = 0; x < sizeof(dscp_pool1) / sizeof(dscp_pool1[0]); x++) {
+ if (dscp_pool1[x].space == (tos >> 2))
+ return dscp_pool1[x].name;
+ }
+ }
+
+ return "unknown";
}
int ast_get_ip(struct sockaddr_in *sin, const char *value)
Modified: trunk/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_iax2.c?rev=15435&r1=15434&r2=15435&view=diff
==============================================================================
--- trunk/channels/chan_iax2.c (original)
+++ trunk/channels/chan_iax2.c Mon Mar 27 21:28:52 2006
@@ -9040,7 +9040,7 @@
tosval = ast_variable_retrieve(cfg, "general", "tos");
if (tosval) {
if (ast_str2tos(tosval, &tos))
- ast_log(LOG_WARNING, "Invalid tos value, should be 'lowdelay', 'throughput', 'reliability', 'mincost', or 'none'\n");
+ ast_log(LOG_WARNING, "Invalid tos value, see doc/iptos.txt for more information.\n");
}
while(v) {
if (!strcasecmp(v->name, "bindport")){
@@ -9190,7 +9190,7 @@
ast_context_create(NULL, regcontext, "IAX2");
} else if (!strcasecmp(v->name, "tos")) {
if (ast_str2tos(v->value, &tos))
- ast_log(LOG_WARNING, "Invalid tos value at line %d, should be 'lowdelay', 'throughput', 'reliability', 'mincost', or 'none'\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid tos value at line %d, see doc/iptos.txt for more information.'\n", v->lineno);
} else if (!strcasecmp(v->name, "accountcode")) {
ast_copy_string(accountcode, v->value, sizeof(accountcode));
} else if (!strcasecmp(v->name, "amaflags")) {
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=15435&r1=15434&r2=15435&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Mar 27 21:28:52 2006
@@ -364,7 +364,9 @@
#define DEFAULT_ALLOWGUEST TRUE
#define DEFAULT_SRVLOOKUP FALSE /*!< Recommended setting is ON */
#define DEFAULT_COMPACTHEADERS FALSE
-#define DEFAULT_TOS FALSE
+#define DEFAULT_TOS_SIP 0 /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
+#define DEFAULT_TOS_AUDIO 0 /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
+#define DEFAULT_TOS_VIDEO 0 /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_ALLOW_EXT_DOM TRUE
#define DEFAULT_REALM "asterisk"
#define DEFAULT_NOTIFYRINGING TRUE
@@ -408,7 +410,9 @@
static int global_allowsubscribe; /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE
the global setting is in globals_flag_page2 */
static int global_mwitime; /*!< Time between MWI checks for peers */
-static int global_tos; /*!< IP Type of service */
+static int global_tos_sip; /*!< IP type of service for SIP packets */
+static int global_tos_audio; /*!< IP type of service for audio RTP packets */
+static int global_tos_video; /*!< IP type of service for video RTP packets */
static int compactheaders; /*!< send compact sip headers */
static int recordhistory; /*!< Record SIP history. Off by default */
static int dumphistory; /*!< Dump history to verbose before destroying SIP dialog */
@@ -3274,9 +3278,9 @@
free(p);
return NULL;
}
- ast_rtp_settos(p->rtp, global_tos);
+ ast_rtp_settos(p->rtp, global_tos_audio);
if (p->vrtp)
- ast_rtp_settos(p->vrtp, global_tos);
+ ast_rtp_settos(p->vrtp, global_tos_video);
p->rtptimeout = global_rtptimeout;
p->rtpholdtimeout = global_rtpholdtimeout;
p->rtpkeepalive = global_rtpkeepalive;
@@ -8449,7 +8453,9 @@
ast_cli(fd, " From: Domain: %s\n", default_fromdomain);
ast_cli(fd, " Record SIP history: %s\n", recordhistory ? "On" : "Off");
ast_cli(fd, " Call Events: %s\n", global_callevents ? "On" : "Off");
- ast_cli(fd, " IP ToS: 0x%x\n", global_tos);
+ ast_cli(fd, " IP ToS SIP: %s\n", ast_tos2str(global_tos_sip));
+ ast_cli(fd, " IP ToS RTP audio: %s\n", ast_tos2str(global_tos_audio));
+ ast_cli(fd, " IP ToS RTP video: %s\n", ast_tos2str(global_tos_video));
#ifdef OSP_SUPPORT
ast_cli(fd, " OSP Support: Yes\n");
#else
@@ -12559,6 +12565,7 @@
int auto_sip_domains = FALSE;
struct sockaddr_in old_bindaddr = bindaddr;
int registry_count = 0, peer_count = 0, user_count = 0;
+ int temp_tos = 0;
struct ast_flags debugflag = {0};
cfg = ast_config_load(config);
@@ -12585,7 +12592,9 @@
outboundproxyip.sin_family = AF_INET; /* Type of address: IPv4 */
ourport = DEFAULT_SIP_PORT;
srvlookup = DEFAULT_SRVLOOKUP;
- global_tos = DEFAULT_TOS;
+ global_tos_sip = DEFAULT_TOS_SIP;
+ global_tos_audio = DEFAULT_TOS_AUDIO;
+ global_tos_video = DEFAULT_TOS_VIDEO;
externhost[0] = '\0'; /* External host name (for behind NAT DynDNS support) */
externexpire = 0; /* Expiration for DNS re-issuing */
externrefresh = 10;
@@ -12808,8 +12817,22 @@
if (sip_register(v->value, v->lineno) == 0)
registry_count++;
} else if (!strcasecmp(v->name, "tos")) {
- if (ast_str2tos(v->value, &global_tos))
- ast_log(LOG_WARNING, "Invalid tos value at line %d, should be 'lowdelay', 'throughput', 'reliability', 'mincost', or 'none'\n", v->lineno);
+ if (!ast_str2tos(v->value, &temp_tos)) {
+ global_tos_sip = temp_tos;
+ global_tos_audio = temp_tos;
+ global_tos_video = temp_tos;
+ ast_log(LOG_WARNING, "tos value at line %d is deprecated. See doc/iptos.txt for more information.", v->lineno);
+ } else
+ ast_log(LOG_WARNING, "Invalid tos value at line %d, See doc/iptos.txt for more information.\n", v->lineno);
+ } else if (!strcasecmp(v->name, "tos_sip")) {
+ if (ast_str2tos(v->value, &global_tos_sip))
+ ast_log(LOG_WARNING, "Invalid tos_sip value at line %d, recommended value is 'cs3'. See doc/iptos.txt.\n", v->lineno);
+ } else if (!strcasecmp(v->name, "tos_audio")) {
+ if (ast_str2tos(v->value, &global_tos_audio))
+ ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, recommended value is 'ef'. See doc/iptos.txt.\n", v->lineno);
+ } else if (!strcasecmp(v->name, "tos_video")) {
+ if (ast_str2tos(v->value, &global_tos_video))
+ ast_log(LOG_WARNING, "Invalid tos_video value at line %d, recommended value is 'af41'. See doc/iptos.txt.\n", v->lineno);
} else if (!strcasecmp(v->name, "bindport")) {
if (sscanf(v->value, "%d", &ourport) == 1) {
bindaddr.sin_port = htons(ourport);
@@ -12922,10 +12945,10 @@
if (option_verbose > 1) {
ast_verbose(VERBOSE_PREFIX_2 "SIP Listening on %s:%d\n",
ast_inet_ntoa(iabuf, sizeof(iabuf), bindaddr.sin_addr), ntohs(bindaddr.sin_port));
- ast_verbose(VERBOSE_PREFIX_2 "Using TOS bits %d\n", global_tos);
+ ast_verbose(VERBOSE_PREFIX_2 "Using SIP TOS: %s\n", ast_tos2str(global_tos_sip));
}
- if (setsockopt(sipsock, IPPROTO_IP, IP_TOS, &global_tos, sizeof(global_tos)))
- ast_log(LOG_WARNING, "Unable to set TOS to %d\n", global_tos);
+ if (setsockopt(sipsock, IPPROTO_IP, IP_TOS, &global_tos_sip, sizeof(global_tos_sip)))
+ ast_log(LOG_WARNING, "Unable to set SIP TOS to %s\n", ast_tos2str(global_tos_sip));
}
}
}
Modified: trunk/channels/iax2-provision.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/iax2-provision.c?rev=15435&r1=15434&r2=15435&view=diff
==============================================================================
--- trunk/channels/iax2-provision.c (original)
+++ trunk/channels/iax2-provision.c Mon Mar 27 21:28:52 2006
@@ -46,6 +46,7 @@
#include "asterisk/md5.h"
#include "asterisk/astdb.h"
#include "asterisk/utils.h"
+#include "asterisk/acl.h"
#include "iax2.h"
#include "iax2-provision.h"
#include "iax2-parser.h"
@@ -328,20 +329,8 @@
} else
ast_log(LOG_WARNING, "Ignoring invalid codec '%s' for '%s' at line %d\n", v->value, s, v->lineno);
} else if (!strcasecmp(v->name, "tos")) {
- if (sscanf(v->value, "%d", &x) == 1)
- cur->tos = x & 0xff;
- else if (!strcasecmp(v->value, "lowdelay"))
- cur->tos = IPTOS_LOWDELAY;
- else if (!strcasecmp(v->value, "throughput"))
- cur->tos = IPTOS_THROUGHPUT;
- else if (!strcasecmp(v->value, "reliability"))
- cur->tos = IPTOS_RELIABILITY;
- else if (!strcasecmp(v->value, "mincost"))
- cur->tos = IPTOS_MINCOST;
- else if (!strcasecmp(v->value, "none"))
- cur->tos = 0;
- else
- ast_log(LOG_WARNING, "Invalid tos value at line %d, should be 'lowdelay', 'throughput', 'reliability', 'mincost', or 'none'\n", v->lineno);
+ if (ast_str2tos(v->value, &cur->tos))
+ ast_log(LOG_WARNING, "Invalid tos value at line %d, see doc/iptos.txt for more information.\n", v->lineno);
} else if (!strcasecmp(v->name, "user")) {
strncpy(cur->user, v->value, sizeof(cur->user) - 1);
if (strcmp(cur->user, v->value))
@@ -453,7 +442,7 @@
ast_cli(fd, "Alternate: %s\n", iax_server(iabuf, sizeof(iabuf), cur->altserver));
ast_cli(fd, "Flags: %s\n", iax_provflags2str(iabuf, sizeof(iabuf), cur->flags));
ast_cli(fd, "Format: %s\n", ast_getformatname(cur->format));
- ast_cli(fd, "TOS: %d\n", cur->tos);
+ ast_cli(fd, "TOS: 0x%x\n", cur->tos);
found++;
}
}
Modified: trunk/configs/iax.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/iax.conf.sample?rev=15435&r1=15434&r2=15435&view=diff
==============================================================================
--- trunk/configs/iax.conf.sample (original)
+++ trunk/configs/iax.conf.sample Mon Mar 27 21:28:52 2006
@@ -207,15 +207,8 @@
;
;authdebug=no
;
-; Finally, you can set values for your TOS bits to help improve
-; performance. Valid values are:
-; lowdelay -- Minimize delay
-; throughput -- Maximize throughput
-; reliability -- Maximize reliability
-; mincost -- Minimize cost
-; none -- No flags
-;
-tos=lowdelay
+; See doc/README.tos for a description of the tos parameters.
+;tos=ef
;
; If regcontext is specified, Asterisk will dynamically create and destroy
; a NoOp priority 1 extension for a given peer who registers or unregisters
Modified: trunk/configs/iaxprov.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/iaxprov.conf.sample?rev=15435&r1=15434&r2=15435&view=diff
==============================================================================
--- trunk/configs/iaxprov.conf.sample (original)
+++ trunk/configs/iaxprov.conf.sample Mon Mar 27 21:28:52 2006
@@ -53,10 +53,8 @@
;
flags=register,heartbeat
;
-; tos is the requested type of service setting and may be one a number or
-; 'lowdelay','throughput','reliability','mincost' or 'none'
-;
-tos=lowdelay
+; See doc/README.tos for a description of this parameter.
+;tos=ef
;
; Example iaxy provisioning
;
Modified: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/sip.conf.sample?rev=15435&r1=15434&r2=15435&view=diff
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Mon Mar 27 21:28:52 2006
@@ -58,8 +58,12 @@
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
-;tos=184 ; Set IP QoS to either a keyword or numeric val
-;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
+
+; See doc/README.tos for a description of these parameters.
+;tos_sip=cs3 ; Sets TOS for SIP packets.
+;tos_audio=ef ; Sets TOS for RTP audio packets.
+;tos_video=af41 ; Sets TOS for RTP video packets.
+
;maxexpiry=3600 ; Max length of incoming registrations/subscriptions we allow (seconds)
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outoing registration
Modified: trunk/include/asterisk/acl.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/acl.h?rev=15435&r1=15434&r2=15435&view=diff
==============================================================================
--- trunk/include/asterisk/acl.h (original)
+++ trunk/include/asterisk/acl.h Mon Mar 27 21:28:52 2006
@@ -1,7 +1,7 @@
/*
* Asterisk -- An open source telephony toolkit.
*
- * Copyright (C) 1999 - 2005, Digium, Inc.
+ * Copyright (C) 1999 - 2006, Digium, Inc.
*
* Mark Spencer <markster at digium.com>
*
@@ -38,16 +38,17 @@
struct ast_ha;
-extern void ast_free_ha(struct ast_ha *ha);
-extern struct ast_ha *ast_append_ha(char *sense, char *stuff, struct ast_ha *path);
-extern int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin);
-extern int ast_get_ip(struct sockaddr_in *sin, const char *value);
-extern int ast_get_ip_or_srv(struct sockaddr_in *sin, const char *value, const char *service);
-extern int ast_ouraddrfor(struct in_addr *them, struct in_addr *us);
-extern int ast_lookup_iface(char *iface, struct in_addr *address);
-extern struct ast_ha *ast_duplicate_ha_list(struct ast_ha *original);
-extern int ast_find_ourip(struct in_addr *ourip, struct sockaddr_in bindaddr);
-extern int ast_str2tos(const char *value, int *tos);
+void ast_free_ha(struct ast_ha *ha);
+struct ast_ha *ast_append_ha(char *sense, char *stuff, struct ast_ha *path);
+int ast_apply_ha(struct ast_ha *ha, struct sockaddr_in *sin);
+int ast_get_ip(struct sockaddr_in *sin, const char *value);
+int ast_get_ip_or_srv(struct sockaddr_in *sin, const char *value, const char *service);
+int ast_ouraddrfor(struct in_addr *them, struct in_addr *us);
+int ast_lookup_iface(char *iface, struct in_addr *address);
+struct ast_ha *ast_duplicate_ha_list(struct ast_ha *original);
+int ast_find_ourip(struct in_addr *ourip, struct sockaddr_in bindaddr);
+int ast_str2tos(const char *value, unsigned int *tos);
+const char *ast_tos2str(unsigned int tos);
#if defined(__cplusplus) || defined(c_plusplus)
}
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