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optimize the 'quick response' code a bit more... no more malloc() or memset() for each response

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expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@45408 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2006-10-17 22:24:10 +00:00
parent 5658b611e5
commit 227d415709
5 changed files with 58 additions and 38 deletions
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52
channels/chan_sip.c
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@@ -722,7 +722,7 @@ struct sip_auth {
#define SIP_PROG_INBAND_NEVER (0 << 25)
#define SIP_PROG_INBAND_NO (1 << 25)
#define SIP_PROG_INBAND_YES (2 << 25)
#define SIP_FREE_BIT (1 << 27) /*!< Undefined bit - not in use */
#define SIP_NO_HISTORY (1 << 27) /*!< Suppress recording request/response history */
#define SIP_CALL_LIMIT (1 << 28) /*!< Call limit enforced for this call */
#define SIP_SENDRPID (1 << 29) /*!< Remote Party-ID Support */
#define SIP_INC_COUNT (1 << 30) /*!< Did this connection increment the counter of in-use calls? */
@@ -1792,7 +1792,7 @@ static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
{
va_list ap;
if (!recordhistory || !p)
if (!p)
return;
va_start(ap, fmt);
append_history_va(p, fmt, ap);
@@ -1979,7 +1979,7 @@ static void sip_scheddestroy(struct sip_pvt *p, int ms)
}
if (sip_debug_test_pvt(p))
ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
if (recordhistory)
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "SchedDestroy", "%d ms", ms);
if (p->autokillid > -1)
@@ -2113,7 +2113,7 @@ static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmitty
ast_inet_ntoa(dst->sin_addr),
ntohs(dst->sin_port), req->data);
}
if (recordhistory) {
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
struct sip_request tmp;
parse_copy(&tmp, req);
append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"),
@@ -2139,7 +2139,7 @@ static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittyp
else
ast_verbose("%sTransmitting (no NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(p->sa.sin_addr), ntohs(p->sa.sin_port), req->data);
}
if (recordhistory) {
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
struct sip_request tmp;
parse_copy(&tmp, req);
append_history(p, reliable ? "TxReqRel" : "TxReq", "%s / %s - %s", tmp.data, get_header(&tmp, "CSeq"), sip_methods[tmp.method].text);
@@ -2810,7 +2810,7 @@ static void sip_registry_destroy(struct sip_registry *reg)
ast_sched_del(sched, reg->expire);
if (reg->timeout > -1)
ast_sched_del(sched, reg->timeout);
ast_string_field_free_all(reg);
ast_string_field_free_pools(reg);
regobjs--;
free(reg);
@@ -2903,7 +2903,7 @@ static void __sip_destroy(struct sip_pvt *p, int lockowner)
}
ast_mutex_destroy(&p->lock);
ast_string_field_free_all(p);
ast_string_field_free_pools(p);
free(p);
}
@@ -3318,7 +3318,7 @@ static int sip_hangup(struct ast_channel *ast)
transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
/* Get RTCP quality before end of call */
if (recordhistory) {
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
if (p->rtp)
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->vrtp)
@@ -3811,7 +3811,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
for (v = i->chanvars ; v ; v = v->next)
pbx_builtin_setvar_helper(tmp,v->name,v->value);
if (recordhistory)
if (!ast_test_flag(&i->flags[0], SIP_NO_HISTORY))
append_history(i, "NewChan", "Channel %s - from %s", tmp->name, i->callid);
return tmp;
@@ -4119,6 +4119,8 @@ static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *si
ast_copy_flags(&p->flags[0], &global_flags[0], SIP_FLAGS_TO_COPY);
ast_copy_flags(&p->flags[1], &global_flags[1], SIP_PAGE2_FLAGS_TO_COPY);
ast_set2_flag(&p->flags[0], !recordhistory, SIP_NO_HISTORY);
p->branch = ast_random();
make_our_tag(p->tag, sizeof(p->tag));
p->ocseq = INITIAL_CSEQ;
@@ -5574,12 +5576,14 @@ static int transmit_response_using_temp(ast_string_field callid, struct sockaddr
return -1;
}
memset(p, 0, sizeof(*p));
/* if the structure was just allocated, initialize it */
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) {
ast_set_flag(&p->flags[0], SIP_NO_HISTORY);
if (ast_string_field_init(p, 512))
return -1;
}
/* Initialize the bare minimum */
if (ast_string_field_init(p, 512))
return -1;
p->method = intended_method;
if (sin) {
@@ -5606,7 +5610,7 @@ static int transmit_response_using_temp(ast_string_field callid, struct sockaddr
/* Use this temporary pvt structure to send the message */
__transmit_response(p, msg, req, XMIT_UNRELIABLE);
/* Now do a simple destruction */
/* Free the string fields, but not the pool space */
ast_string_field_free_all(p);
return 0;
@@ -6317,7 +6321,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if (sipdebug)
add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
if (recordhistory)
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "ReInv", "Re-invite sent");
add_sdp(&req, p);
/* Use this as the basis */
@@ -6942,7 +6946,7 @@ static int sip_reregister(void *data)
if (!r)
return 0;
if (r->call && recordhistory)
if (r->call && !ast_test_flag(&r->call->flags[0], SIP_NO_HISTORY))
append_history(r->call, "RegistryRenew", "Account: %s@%s", r->username, r->hostname);
/* Since registry's are only added/removed by the the monitor thread, this
may be overkill to reference/dereference at all here */
@@ -7042,7 +7046,7 @@ static int transmit_register(struct sip_registry *r, int sipmethod, const char *
ast_log(LOG_WARNING, "Unable to allocate registration transaction (memory or socket error)\n");
return 0;
}
if (recordhistory)
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "RegistryInit", "Account: %s@%s", r->username, r->hostname);
/* Find address to hostname */
if (create_addr(p, r->hostname)) {
@@ -10786,7 +10790,7 @@ static int do_register_auth(struct sip_pvt *p, struct sip_request *req, char *he
/* No old challenge */
return -1;
}
if (recordhistory)
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "RegistryAuth", "Try: %d", p->authtries);
if (sip_debug_test_pvt(p) && p->registry)
ast_verbose("Responding to challenge, registration to domain/host name %s\n", p->registry->hostname);
@@ -12992,7 +12996,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (option_debug)
ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
}
if (recordhistory) /* This is a response, note what it was for */
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a response, note what it was for */
append_history(p, "ReInv", "Re-invite received");
}
} else if (debug)
@@ -13791,18 +13795,18 @@ static int handle_request_bye(struct sip_pvt *p, struct sip_request *req)
ast_set_flag(&p->flags[0], SIP_ALREADYGONE);
/* Get RTCP quality before end of call */
if (recordhistory || p->owner) {
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY) || p->owner) {
char *audioqos, *videoqos;
if (p->rtp) {
audioqos = ast_rtp_get_quality(p->rtp);
if (recordhistory)
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "RTCPaudio", "Quality:%s", audioqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPAUDIOQOS", audioqos);
}
if (p->vrtp) {
videoqos = ast_rtp_get_quality(p->vrtp);
if (recordhistory)
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "RTCPvideo", "Quality:%s", videoqos);
if (p->owner)
pbx_builtin_setvar_helper(p->owner, "RTPVIDEOQOS", videoqos);
@@ -14429,7 +14433,7 @@ static int sipsock_read(int *id, int fd, short events, void *ignore)
}
p->recv = sin;
if (recordhistory) /* This is a request or response, note what it was for */
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY)) /* This is a request or response, note what it was for */
append_history(p, "Rx", "%s / %s / %s", req.data, get_header(&req, "CSeq"), req.rlPart2);
if (!lockretry) {
@@ -16438,7 +16442,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
}
if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (chan->_state != AST_STATE_UP) { /* We are in early state */
if (recordhistory)
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
if (option_debug)
ast_log(LOG_DEBUG, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(rtp ? p->redirip.sin_addr : p->ourip));