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authorOlle Johansson <oej@edvina.net>2007-12-16 10:51:53 +0000
committerOlle Johansson <oej@edvina.net>2007-12-16 10:51:53 +0000
commit17afebc1a66f6cb114abfbd0a0490f7e45b3bdc6 (patch)
tree1737bbf754fc80795d7da8a7e44ef757640a050f /channels
parent4d0ceb96ba5d97407f0ca819b0053055c4cc9d5d (diff)
HUGE improvements to QoS/CoS handling by IgorG
- Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_h323.c20
-rw-r--r--channels/chan_iax2.c8
-rw-r--r--channels/chan_mgcp.c19
-rw-r--r--channels/chan_sip.c30
-rw-r--r--channels/chan_skinny.c29
-rw-r--r--channels/chan_unistim.c24
-rw-r--r--channels/iax2-provision.c2
7 files changed, 102 insertions, 30 deletions
diff --git a/channels/chan_h323.c b/channels/chan_h323.c
index f6f138c87..87fbcd786 100644
--- a/channels/chan_h323.c
+++ b/channels/chan_h323.c
@@ -970,7 +970,7 @@ static int __oh323_rtp_create(struct oh323_pvt *pvt)
if (h323debug)
ast_debug(1, "Created RTP channel\n");
- ast_rtp_setqos(pvt->rtp, tos, cos);
+ ast_rtp_setqos(pvt->rtp, tos, cos, "H323 RTP");
if (h323debug)
ast_debug(1, "Setting NAT on RTP to %d\n", pvt->options.nat);
@@ -2904,13 +2904,23 @@ static int reload_config(int is_reload)
} else {
memcpy(&bindaddr.sin_addr, hp->h_addr, sizeof(bindaddr.sin_addr));
}
- } else if (!strcasecmp(v->name, "tos")) {
+ } else if (!strcasecmp(v->name, "tos")) { /* Needs to be removed in next release */
+ ast_log(LOG_WARNING, "The \"tos\" setting is deprecated in this version of Asterisk. Please change to \"tos_audio\".\n");
if (ast_str2tos(v->value, &tos)) {
- ast_log(LOG_WARNING, "Invalid tos value at line %d, for more info read doc/qos.tex\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
}
- } else if (!strcasecmp(v->name, "cos")) {
+ } else if (!strcasecmp(v->name, "tos_audio")) {
+ if (ast_str2tos(v->value, &tos)) {
+ ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
+ }
+ } else if (!strcasecmp(v->name, "cos")) {
+ ast_log(LOG_WARNING, "The \"cos\" setting is deprecated in this version of Asterisk. Please change to \"cos_audio\".\n");
+ if (ast_str2cos(v->value, &cos)) {
+ ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
+ }
+ } else if (!strcasecmp(v->name, "cos_audio")) {
if (ast_str2cos(v->value, &cos)) {
- ast_log(LOG_WARNING, "Invalid cos value at line %d, for more info read doc/qos.tex\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
}
} else if (!strcasecmp(v->name, "gatekeeper")) {
if (!strcasecmp(v->value, "DISABLE")) {
diff --git a/channels/chan_iax2.c b/channels/chan_iax2.c
index 3288800e7..b1205ef92 100644
--- a/channels/chan_iax2.c
+++ b/channels/chan_iax2.c
@@ -10422,13 +10422,13 @@ static int set_config(char *config_file, int reload)
tosval = ast_variable_retrieve(cfg, "general", "tos");
if (tosval) {
if (ast_str2tos(tosval, &tos))
- ast_log(LOG_WARNING, "Invalid tos value, see doc/qos.tex for more information.\n");
+ ast_log(LOG_WARNING, "Invalid tos value, refer to QoS documentation\n");
}
/* Seed initial cos value */
tosval = ast_variable_retrieve(cfg, "general", "cos");
if (tosval) {
if (ast_str2cos(tosval, &cos))
- ast_log(LOG_WARNING, "Invalid cos value, see doc/qos.tex for more information.\n");
+ ast_log(LOG_WARNING, "Invalid cos value, refer to QoS documentation\n");
}
while(v) {
if (!strcasecmp(v->name, "bindport")){
@@ -10601,10 +10601,10 @@ static int set_config(char *config_file, int reload)
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, see doc/qos.tex for more information.'\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid tos value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos")) {
if (ast_str2cos(v->value, &cos))
- ast_log(LOG_WARNING, "Invalid cos value at line %d, see doc/qos.tex for more information.'\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid cos value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "accountcode")) {
ast_copy_string(accountcode, v->value, sizeof(accountcode));
} else if (!strcasecmp(v->name, "mohinterpret")) {
diff --git a/channels/chan_mgcp.c b/channels/chan_mgcp.c
index ae7a3ed14..bd2c66d62 100644
--- a/channels/chan_mgcp.c
+++ b/channels/chan_mgcp.c
@@ -153,8 +153,9 @@ static ast_group_t cur_callergroup = 0;
static ast_group_t cur_pickupgroup = 0;
static unsigned int tos = 0;
-
+static unsigned int tos_audio = 0;
static unsigned int cos = 0;
+static unsigned int cos_audio = 0;
static int immediate = 0;
@@ -2591,8 +2592,10 @@ static void start_rtp(struct mgcp_subchannel *sub)
sub->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
if (sub->rtp && sub->owner)
ast_channel_set_fd(sub->owner, 0, ast_rtp_fd(sub->rtp));
- if (sub->rtp)
+ if (sub->rtp) {
+ ast_rtp_setqos(sub->rtp, tos_audio, cos_audio, "MGCP RTP");
ast_rtp_setnat(sub->rtp, sub->nat);
+ }
#if 0
ast_rtp_set_callback(p->rtp, rtpready);
ast_rtp_set_data(p->rtp, p);
@@ -4097,10 +4100,16 @@ static int reload_config(int reload)
capability &= ~format;
} else if (!strcasecmp(v->name, "tos")) {
if (ast_str2tos(v->value, &tos))
- ast_log(LOG_WARNING, "Invalid tos value at line %d, see doc/qos.tex for more information.\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid tos value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "tos_audio")) {
+ if (ast_str2tos(v->value, &tos_audio))
+ ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos")) {
if (ast_str2cos(v->value, &cos))
- ast_log(LOG_WARNING, "Invalid cos value at line %d, see doc/qos.tex for more information.\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid cos value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "cos_audio")) {
+ if (ast_str2cos(v->value, &cos_audio))
+ ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "port")) {
if (sscanf(v->value, "%d", &ourport) == 1) {
bindaddr.sin_port = htons(ourport);
@@ -4184,7 +4193,7 @@ static int reload_config(int reload)
} else {
ast_verb(2, "MGCP Listening on %s:%d\n",
ast_inet_ntoa(bindaddr.sin_addr), ntohs(bindaddr.sin_port));
- ast_netsock_set_qos(mgcpsock, tos, cos);
+ ast_netsock_set_qos(mgcpsock, tos, cos, "MGCP");
}
}
ast_mutex_unlock(&netlock);
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 57b17f198..6e5a3b17a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -545,7 +545,7 @@ static const struct cfsip_options {
#define DEFAULT_COS_SIP 4
#define DEFAULT_COS_AUDIO 5
#define DEFAULT_COS_VIDEO 6
-#define DEFAULT_COS_TEXT 0
+#define DEFAULT_COS_TEXT 5
#define DEFAULT_ALLOW_EXT_DOM TRUE
#define DEFAULT_REALM "asterisk"
#define DEFAULT_NOTIFYRINGING TRUE
@@ -5130,14 +5130,14 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
ast_free(p);
return NULL;
}
- ast_rtp_setqos(p->rtp, global_tos_audio, global_cos_audio);
+ ast_rtp_setqos(p->rtp, global_tos_audio, global_cos_audio, "SIP RTP");
ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
ast_rtp_set_rtptimeout(p->rtp, global_rtptimeout);
ast_rtp_set_rtpholdtimeout(p->rtp, global_rtpholdtimeout);
ast_rtp_set_rtpkeepalive(p->rtp, global_rtpkeepalive);
if (p->vrtp) {
- ast_rtp_setqos(p->vrtp, global_tos_video, global_cos_video);
+ ast_rtp_setqos(p->vrtp, global_tos_video, global_cos_video, "SIP VRTP");
ast_rtp_setdtmf(p->vrtp, 0);
ast_rtp_setdtmfcompensate(p->vrtp, 0);
ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout);
@@ -5145,7 +5145,7 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive);
}
if (p->trtp) {
- ast_rtp_setqos(p->trtp, global_tos_text, global_cos_text);
+ ast_rtp_setqos(p->trtp, global_tos_text, global_cos_text, "SIP TRTP");
ast_rtp_setdtmf(p->trtp, 0);
ast_rtp_setdtmfcompensate(p->trtp, 0);
}
@@ -18575,24 +18575,28 @@ static int reload_config(enum channelreloadreason reason)
registry_count++;
} 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/qos.tex.\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid tos_sip value at line %d, refer to QoS documentation\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/qos.tex.\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\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/qos.tex.\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid tos_video value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "tos_text")) {
if (ast_str2tos(v->value, &global_tos_text))
- ast_log(LOG_WARNING, "Invalid tos_text value at line %d, recommended value is 'af41'. See doc/qos.tex.\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid tos_text value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos_sip")) {
- ast_str2cos(v->value, &global_cos_sip);
+ if (ast_str2cos(v->value, &global_cos_sip))
+ ast_log(LOG_WARNING, "Invalid cos_sip value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos_audio")) {
- ast_str2cos(v->value, &global_cos_audio);
+ if (ast_str2cos(v->value, &global_cos_audio))
+ ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos_video")) {
- ast_str2cos(v->value, &global_cos_video);
+ if (ast_str2cos(v->value, &global_cos_video))
+ ast_log(LOG_WARNING, "Invalid cos_video value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "cos_text")) {
- ast_str2cos(v->value, &global_cos_text);
+ if (ast_str2cos(v->value, &global_cos_text))
+ ast_log(LOG_WARNING, "Invalid cos_text value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "bindport")) {
int i;
if (sscanf(v->value, "%d", &i) == 1) {
@@ -18761,7 +18765,7 @@ static int reload_config(enum channelreloadreason reason)
} else {
ast_verb(2, "SIP Listening on %s:%d\n",
ast_inet_ntoa(bindaddr.sin_addr), ntohs(bindaddr.sin_port));
- ast_netsock_set_qos(sipsock, global_tos_sip, global_cos_sip);
+ ast_netsock_set_qos(sipsock, global_tos_sip, global_cos_sip, "SIP");
}
}
}
diff --git a/channels/chan_skinny.c b/channels/chan_skinny.c
index 5c821e183..6c526d461 100644
--- a/channels/chan_skinny.c
+++ b/channels/chan_skinny.c
@@ -50,6 +50,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
+#include "asterisk/netsock.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
@@ -90,6 +91,13 @@ enum skinny_codecs {
#define DEFAULT_SKINNY_BACKLOG 2
#define SKINNY_MAX_PACKET 1000
+static unsigned int tos = 0;
+static unsigned int tos_audio = 0;
+static unsigned int tos_video = 0;
+static unsigned int cos = 0;
+static unsigned int cos_audio = 0;
+static unsigned int cos_video = 0;
+
static int keep_alive = 120;
static char vmexten[AST_MAX_EXTENSION]; /* Voicemail pilot number */
static char used_context[AST_MAX_EXTENSION]; /* Voicemail pilot number */
@@ -2976,9 +2984,11 @@ static void start_rtp(struct skinny_subchannel *sub)
ast_channel_set_fd(sub->owner, 3, ast_rtcp_fd(sub->vrtp));
}
if (sub->rtp) {
+ ast_rtp_setqos(sub->rtp, tos_audio, cos_audio, "Skinny RTP");
ast_rtp_setnat(sub->rtp, l->nat);
}
if (sub->vrtp) {
+ ast_rtp_setqos(sub->vrtp, tos_video, cos_video, "Skinny VRTP");
ast_rtp_setnat(sub->vrtp, l->nat);
}
/* Set Frame packetization */
@@ -5516,6 +5526,24 @@ static int reload_config(void)
ast_copy_string(regcontext, v->value, sizeof(regcontext));
} else if (!strcasecmp(v->name, "dateformat")) {
memcpy(date_format, v->value, sizeof(date_format));
+ } else if (!strcasecmp(v->name, "tos")) {
+ if (ast_str2tos(v->value, &tos))
+ ast_log(LOG_WARNING, "Invalid tos value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "tos_audio")) {
+ if (ast_str2tos(v->value, &tos_audio))
+ ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "tos_video")) {
+ if (ast_str2tos(v->value, &tos_video))
+ ast_log(LOG_WARNING, "Invalid tos_video value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "cos")) {
+ if (ast_str2cos(v->value, &cos))
+ ast_log(LOG_WARNING, "Invalid cos value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "cos_audio")) {
+ if (ast_str2cos(v->value, &cos_audio))
+ ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "cos_video")) {
+ if (ast_str2cos(v->value, &cos_video))
+ ast_log(LOG_WARNING, "Invalid cos_video value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "allow")) {
ast_parse_allow_disallow(&default_prefs, &default_capability, v->value, 1);
} else if (!strcasecmp(v->name, "disallow")) {
@@ -5604,6 +5632,7 @@ static int reload_config(void)
}
ast_verb(2, "Skinny listening on %s:%d\n",
ast_inet_ntoa(bindaddr.sin_addr), ntohs(bindaddr.sin_port));
+ ast_netsock_set_qos(skinnysock, tos, cos, "Skinny");
ast_pthread_create_background(&accept_t,NULL, accept_thread, NULL);
}
}
diff --git a/channels/chan_unistim.c b/channels/chan_unistim.c
index 9c1a360ad..d4cb0f347 100644
--- a/channels/chan_unistim.c
+++ b/channels/chan_unistim.c
@@ -59,6 +59,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
+#include "asterisk/netsock.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
@@ -214,6 +215,10 @@ static int unistim_port;
static enum autoprovision autoprovisioning = AUTOPROVISIONING_NO;
static int unistim_keepalive;
static int unistimsock = -1;
+static unsigned int tos = 0;
+static unsigned int tos_audio = 0;
+static unsigned int cos = 0;
+static unsigned int cos_audio = 0;
static struct io_context *io;
static struct sched_context *sched;
static struct sockaddr_in public_ip = { 0, };
@@ -2075,8 +2080,10 @@ static void start_rtp(struct unistim_subchannel *sub)
sub->owner->fds[0] = ast_rtp_fd(sub->rtp);
sub->owner->fds[1] = ast_rtcp_fd(sub->rtp);
}
- if (sub->rtp)
+ if (sub->rtp) {
+ ast_rtp_setqos(sub->rtp, tos_audio, cos_audio, "UNISTIM RTP");
ast_rtp_setnat(sub->rtp, sub->parent->parent->nat);
+ }
/* Create the RTP connection */
ast_rtp_get_us(sub->rtp, &us);
@@ -5330,7 +5337,19 @@ static int reload_config(void)
unistim_keepalive = atoi(v->value);
else if (!strcasecmp(v->name, "port"))
unistim_port = atoi(v->value);
- else if (!strcasecmp(v->name, "autoprovisioning")) {
+ else if (!strcasecmp(v->name, "tos")) {
+ if (ast_str2tos(v->value, &tos))
+ ast_log(LOG_WARNING, "Invalid tos value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "tos_audio")) {
+ if (ast_str2tos(v->value, &tos_audio))
+ ast_log(LOG_WARNING, "Invalid tos_audio value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "cos")) {
+ if (ast_str2cos(v->value, &cos))
+ ast_log(LOG_WARNING, "Invalid cos value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "cos_audio")) {
+ if (ast_str2cos(v->value, &cos_audio))
+ ast_log(LOG_WARNING, "Invalid cos_audio value at line %d, refer to QoS documentation\n", v->lineno);
+ } else if (!strcasecmp(v->name, "autoprovisioning")) {
if (!strcasecmp(v->value, "no"))
autoprovisioning = AUTOPROVISIONING_NO;
else if (!strcasecmp(v->value, "yes"))
@@ -5511,6 +5530,7 @@ static int reload_config(void)
"UNISTIM Listening on %s:%d\n",
ast_inet_ntoa(bindaddr.sin_addr), htons(bindaddr.sin_port));
}
+ ast_netsock_set_qos(unistimsock, tos, cos, "UNISTIM");
}
return 0;
}
diff --git a/channels/iax2-provision.c b/channels/iax2-provision.c
index 3ddec3e06..5b52a0934 100644
--- a/channels/iax2-provision.c
+++ b/channels/iax2-provision.c
@@ -323,7 +323,7 @@ static int iax_template_parse(struct iax_template *cur, struct ast_config *cfg,
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 (ast_str2tos(v->value, &cur->tos))
- ast_log(LOG_WARNING, "Invalid tos value at line %d, see doc/qos.tex for more information.\n", v->lineno);
+ ast_log(LOG_WARNING, "Invalid tos value at line %d, refer to QoS documentation\n", v->lineno);
} else if (!strcasecmp(v->name, "user")) {
strncpy(cur->user, v->value, sizeof(cur->user) - 1);
if (strcmp(cur->user, v->value))