[svn-commits] oej: branch oej/ringstatelimit-1.4 r179738 - /team/oej/ringstatelimit-1.4/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Tue Mar 3 09:48:55 CST 2009
Author: oej
Date: Tue Mar 3 09:48:51 2009
New Revision: 179738
URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=179738
Log:
Adding a diff file of the current state to make testing simple
Added:
team/oej/ringstatelimit-1.4/ringstatelimit-1.4.diff (with props)
Added: team/oej/ringstatelimit-1.4/ringstatelimit-1.4.diff
URL: http://svn.digium.com/svn-view/asterisk/team/oej/ringstatelimit-1.4/ringstatelimit-1.4.diff?view=auto&rev=179738
==============================================================================
--- team/oej/ringstatelimit-1.4/ringstatelimit-1.4.diff (added)
+++ team/oej/ringstatelimit-1.4/ringstatelimit-1.4.diff Tue Mar 3 09:48:51 2009
@@ -1,0 +1,154 @@
+--- ../asterisk-1.4/channels/chan_sip.c 2009-03-03 12:34:26.000000000 +0100
++++ channels/chan_sip.c 2009-03-03 16:43:47.000000000 +0100
+@@ -512,6 +512,7 @@
+ #define DEFAULT_AUTOCREATEPEER FALSE
+ #define DEFAULT_QUALIFY FALSE
+ #define DEFAULT_T1MIN 100 /*!< 100 MS for minimal roundtrip time */
++#define DEFAULT_TIMERB SIP_TRANS_TIMEOUT /*!< Default timeout for an invite in Call State - 64*T1 */
+ #define DEFAULT_MAX_CALL_BITRATE (384) /*!< Max bitrate for video */
+ #ifndef DEFAULT_USERAGENT
+ #define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
+@@ -566,6 +567,7 @@
+ static int allow_external_domains; /*!< Accept calls to external SIP domains? */
+ static int global_callevents; /*!< Whether we send manager events or not */
+ static int global_t1min; /*!< T1 roundtrip time minimum */
++static int global_timer_b; /*!< Timeout in ring/call state (default: 64 * T1) */
+ static int global_autoframing; /*!< Turn autoframing on or off. */
+ static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */
+
+@@ -961,6 +963,7 @@
+ int lastnoninvite; /*!< Last Cseq of non-invite */
+ struct ast_flags flags[2]; /*!< SIP_ flags */
+ int timer_t1; /*!< SIP timer T1, ms rtt */
++ int timer_b; /*!< SIP timer B, call state timeout */
+ unsigned int sipoptions; /*!< Supported SIP options on the other end */
+ struct ast_codec_pref prefs; /*!< codec prefs */
+ int capability; /*!< Special capability (codec) */
+@@ -1048,6 +1051,7 @@
+ int retransid; /*!< Retransmission ID */
+ int timer_a; /*!< SIP timer A, retransmission timer */
+ int timer_t1; /*!< SIP Timer T1, estimated RTT or 500 ms */
++ int timer_b; /*!< SIP Timer B, call state timeout */
+ int packetlen; /*!< Length of packet */
+ char data[0];
+ };
+@@ -1138,6 +1142,7 @@
+ int maxms; /*!< Max ms we will accept for the host to be up, 0 to not monitor */
+ struct timeval ps; /*!< Ping send time */
+
++ int timer_b; /*!< SIP timer B, call state timeout */
+ struct sockaddr_in defaddr; /*!< Default IP address, used until registration */
+ struct ast_ha *ha; /*!< Access control list */
+ struct ast_ha *contactha; /*!< Restrict what IPs are allowed in the Contact header (for registration) */
+@@ -2145,8 +2150,10 @@
+ static void sip_scheddestroy(struct sip_pvt *p, int ms)
+ {
+ if (ms < 0) {
+- if (p->timer_t1 == 0)
++ if (p->timer_t1 == 0) {
+ p->timer_t1 = 500; /* Set timer T1 if not set (RFC 3261) */
++ p->timer_b = global_timer_b;
++ }
+ ms = p->timer_t1 * 64;
+ }
+ if (sip_debug_test_pvt(p))
+@@ -2902,6 +2909,12 @@
+ /* Minimum is settable or default to 100 ms */
+ if (peer->maxms && peer->lastms)
+ dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
++ if (peer->timer_b) {
++ dialog->timer_b = peer->timer_b;
++ } else {
++ dialog->timer_b = 64 * dialog->timer_t1;
++ }
++
+ if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
+ (ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
+ dialog->noncodeccapability |= AST_RTP_DTMF;
+@@ -2936,6 +2949,7 @@
+ *port++ = '\0';
+ dialog->sa.sin_family = AF_INET;
+ dialog->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
++ dialog->timer_b = global_timer_b; /* Default SIP transaction timer B (RFC 3261) */
+ p = find_peer(peer, NULL, 1, 0);
+
+ if (p) {
+@@ -3098,7 +3112,7 @@
+
+ /* Initialize auto-congest time */
+ AST_SCHED_DEL(sched, p->initid);
+- p->initid = ast_sched_add(sched, p->maxtime ? (p->maxtime * 4) : SIP_TRANS_TIMEOUT, auto_congest, p);
++ p->initid = ast_sched_add(sched, p->timer_b, auto_congest, p);
+ }
+ } else {
+ ast->hangupcause = AST_CAUSE_USER_BUSY;
+@@ -4547,8 +4561,10 @@
+ p->stateid = -1;
+ p->prefs = default_prefs; /* Set default codecs for this call */
+
+- if (intended_method != SIP_OPTIONS) /* Peerpoke has it's own system */
++ if (intended_method != SIP_OPTIONS) { /* Peerpoke has it's own system */
+ p->timer_t1 = 500; /* Default SIP retransmission timer T1 (RFC 3261) */
++ p->timer_b = global_timer_b;
++ }
+
+ if (sin) {
+ p->sa = *sin;
+@@ -9839,6 +9855,10 @@
+ ast_string_field_set(p, cid_num, tmp);
+ }
+ do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT_ROUTE));
++ if (peer->timer_b)
++ p->timer_b = peer->timer_b;
++ else
++ p->timer_b = 64 * 500;
+
+ ast_string_field_set(p, peersecret, peer->secret);
+ ast_string_field_set(p, peermd5secret, peer->md5secret);
+@@ -11009,7 +11029,8 @@
+ ast_cli(fd, " Codec Order: ");
+ print_codec_to_cli(fd, &default_prefs);
+ ast_cli(fd, "\n");
+- ast_cli(fd, " T1 minimum: %d\n", global_t1min);
++ ast_cli(fd, " Timer T1 minimum: %d\n", global_t1min);
++ ast_cli(fd, " Timer B: %d\n", global_timer_b);
+ ast_cli(fd, " Relax DTMF: %s\n", global_relaxdtmf ? "Yes" : "No");
+ ast_cli(fd, " Compact SIP headers: %s\n", compactheaders ? "Yes" : "No");
+ ast_cli(fd, " RTP Keepalive: %d %s\n", global_rtpkeepalive, global_rtpkeepalive ? "" : "(Disabled)" );
+@@ -17327,6 +17348,7 @@
+ peer->pickupgroup = 0;
+ peer->maxms = default_qualify;
+ peer->prefs = default_prefs;
++ peer->timer_b = global_timer_b;
+ }
+
+ /*! \brief Create temporary peer (used in autocreatepeer mode) */
+@@ -17523,6 +17545,11 @@
+ peer->call_limit = atoi(v->value);
+ if (peer->call_limit < 0)
+ peer->call_limit = 0;
++ } else if (!strcasecmp(v->name, "timerb")) {
++ if ((sscanf(v->value, "%d", &peer->timer_b) != 1) || (peer->timer_b < 0)) {
++ ast_log(LOG_WARNING, "'%s' is not a valid Timer B time at line %d. Using default.\n", v->value, v->lineno);
++ peer->timer_b = global_timer_b;
++ }
+ } else if (!strcasecmp(v->name, "amaflags")) {
+ format = ast_cdr_amaflags2int(v->value);
+ if (format < 0) {
+@@ -17799,6 +17826,7 @@
+ global_relaxdtmf = FALSE;
+ global_callevents = FALSE;
+ global_t1min = DEFAULT_T1MIN;
++ global_timer_b = DEFAULT_TIMERB;
+
+ global_matchexterniplocally = FALSE;
+
+@@ -17840,6 +17868,8 @@
+ ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_IGNOREREGEXPIRE);
+ } else if (!strcasecmp(v->name, "t1min")) {
+ global_t1min = atoi(v->value);
++ } else if (!strcasecmp(v->name, "timerb")) {
++ global_timer_b = atoi(v->value);
+ } else if (!strcasecmp(v->name, "dynamic_exclude_static") || !strcasecmp(v->name, "dynamic_excludes_static")) {
+ global_dynamic_exclude_static = ast_true(v->value);
+ } else if (!strcasecmp(v->name, "contactpermit") || !strcasecmp(v->name, "contactdeny")) {
Propchange: team/oej/ringstatelimit-1.4/ringstatelimit-1.4.diff
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Propchange: team/oej/ringstatelimit-1.4/ringstatelimit-1.4.diff
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Propchange: team/oej/ringstatelimit-1.4/ringstatelimit-1.4.diff
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