Index: channels/chan_sip.c =================================================================== RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v retrieving revision 1.784 diff -u -r1.784 chan_sip.c --- channels/chan_sip.c 20 Jul 2005 17:05:18 -0000 1.784 +++ channels/chan_sip.c 21 Jul 2005 20:32:17 -0000 @@ -123,21 +123,21 @@ static const char config[] = "sip.conf"; static const char notify_config[] = "sip_notify.conf"; -#define SIP_REGISTER 1 -#define SIP_OPTIONS 2 -#define SIP_NOTIFY 3 -#define SIP_INVITE 4 -#define SIP_ACK 5 -#define SIP_PRACK 6 -#define SIP_BYE 7 -#define SIP_REFER 8 -#define SIP_SUBSCRIBE 9 -#define SIP_MESSAGE 10 -#define SIP_UPDATE 11 -#define SIP_INFO 12 -#define SIP_CANCEL 13 -#define SIP_PUBLISH 14 -#define SIP_RESPONSE 100 +#define SIP_RESPONSE 1 +#define SIP_REGISTER 2 +#define SIP_OPTIONS 3 +#define SIP_NOTIFY 4 +#define SIP_INVITE 5 +#define SIP_ACK 6 +#define SIP_PRACK 7 +#define SIP_BYE 8 +#define SIP_REFER 9 +#define SIP_SUBSCRIBE 10 +#define SIP_MESSAGE 12 +#define SIP_UPDATE 13 +#define SIP_INFO 14 +#define SIP_CANCEL 15 +#define SIP_PUBLISH 16 #define RTP 1 #define NO_RTP 0 @@ -147,6 +147,7 @@ char *text; } sip_methods[] = { { 0, RTP, "-UNKNOWN-" }, + { SIP_RESPONSE, NO_RTP, "SIP/2.0" }, { SIP_REGISTER, NO_RTP, "REGISTER" }, { SIP_OPTIONS, NO_RTP, "OPTIONS" }, { SIP_NOTIFY, NO_RTP, "NOTIFY" }, @@ -384,6 +385,7 @@ int lines; /* SDP Content */ char *line[SIP_MAX_LINES]; char data[SIP_MAX_PACKET]; + int debug; /* Debug flag for this packet */ }; struct sip_pkt; @@ -586,6 +588,7 @@ struct sip_pkt { struct sip_pkt *next; /* Next packet */ int retrans; /* Retransmission number */ + int method; /* SIP method for this packet */ int seqno; /* Sequence number */ unsigned int flags; /* non-zero if this is a response packet (e.g. 200 OK) */ struct sip_pvt *owner; /* Owner call */ @@ -787,7 +790,7 @@ static int expire_register(void *data); static int callevents = 0; -static struct ast_channel *sip_request(const char *type, int format, void *data, int *cause); +static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause); static int sip_devicestate(void *data); static int sip_sendtext(struct ast_channel *ast, const char *text); static int sip_call(struct ast_channel *ast, char *dest, int timeout); @@ -803,6 +806,7 @@ static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno); /* Add realm authentication in list */ static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, char *realm); /* Find authentication for a specific realm */ static void append_date(struct sip_request *req); /* Append date to SIP packet */ +static int determine_firstline_parts(struct sip_request *req); /* Definition of this channel for channel registration */ static const struct ast_channel_tech sip_tech = { @@ -810,7 +814,7 @@ .description = "Session Initiation Protocol (SIP)", .capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1), .properties = AST_CHAN_TP_WANTSJITTER, - .requester = sip_request, + .requester = sip_request_call, .devicestate = sip_devicestate, .call = sip_call, .hangup = sip_hangup, @@ -830,8 +834,13 @@ int find_sip_method(char *msg) { int i, res = 0; + + if (!msg || ast_strlen_zero(msg)) + return 0; + /* Strictly speaking, SIP methods are case SENSITIVE, but we don't check */ - for (i=1;(i < (sizeof(sip_methods) / sizeof(sip_methods[0]))) && !res; i++) { + /* following Jon Postel's rule: Be gentle in what you accept, strict with what you send */ + for (i=1; (i < (sizeof(sip_methods) / sizeof(sip_methods[0]))) && !res; i++) { if (!strcasecmp(sip_methods[i].text, msg)) res = sip_methods[i].id; } @@ -1089,7 +1098,7 @@ } /*--- __sip_reliable_xmit: transmit packet with retransmits ---*/ -static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal) +static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod) { struct sip_pkt *pkt; pkt = malloc(sizeof(struct sip_pkt) + len + 1); @@ -1097,6 +1106,7 @@ return -1; memset(pkt, 0, sizeof(struct sip_pkt)); memcpy(pkt->data, data, len); + pkt->method = sipmethod; pkt->packetlen = len; pkt->next = p->packets; pkt->owner = p; @@ -1109,8 +1119,9 @@ pkt->retransid = ast_sched_add(sched, DEFAULT_RETRANS, retrans_pkt, pkt); pkt->next = p->packets; p->packets = pkt; - __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); - if (!strncasecmp(pkt->data, "INVITE", 6)) { + + __sip_xmit(pkt->owner, pkt->data, pkt->packetlen); /* Send packet */ + if (sipmethod == SIP_INVITE) { /* Note this is a pending invite */ p->pendinginvite = seqno; } @@ -1244,7 +1255,7 @@ return res; } -static void parse(struct sip_request *req); +static void parse_request(struct sip_request *req); static char *get_header(struct sip_request *req, char *name); static void copy_request(struct sip_request *dst,struct sip_request *src); @@ -1254,7 +1265,7 @@ memset(dst, 0, sizeof(*dst)); memcpy(dst->data, src->data, sizeof(dst->data)); dst->len = src->len; - parse(dst); + parse_request(dst); } /*--- send_response: Transmit response on SIP request---*/ @@ -1264,6 +1275,7 @@ char iabuf[INET_ADDRSTRLEN]; struct sip_request tmp; char tmpmsg[80]; + if (sip_debug_test_pvt(p)) { if (ast_test_flag(p, SIP_NAT) & SIP_NAT_ROUTE) ast_verbose("%sTransmitting (NAT) to %s:%d:\n%s\n---\n", reliable ? "Reliably " : "", ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port), req->data); @@ -1276,7 +1288,7 @@ snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); append_history(p, "TxRespRel", tmpmsg); } - res = __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1)); + res = __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1), req->method); } else { if (recordhistory) { parse_copy(&tmp, req); @@ -1310,7 +1322,7 @@ snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq")); append_history(p, "TxReqRel", tmpmsg); } - res = __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1)); + res = __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1), req->method); } else { if (recordhistory) { parse_copy(&tmp, req); @@ -2877,27 +2889,18 @@ struct sip_pvt *p; char *callid; char tmp[256] = ""; - char iabuf[INET_ADDRSTRLEN]; - char *cmd; char *tag = "", *c; callid = get_header(req, "Call-ID"); if (pedanticsipchecking) { - /* In principle Call-ID's uniquely identify a call, however some vendors - (i.e. Pingtel) send multiple calls with the same Call-ID and different - tags in order to simplify billing. The RFC does state that we have to - compare tags in addition to the call-id, but this generate substantially - more overhead which is totally unnecessary for the vast majority of sane - SIP implementations, and thus Asterisk does not enable this behavior - by default. Short version: You'll need this option to support conferencing - on the pingtel */ - ast_copy_string(tmp, req->header[0], sizeof(tmp)); - cmd = tmp; - c = strchr(tmp, ' '); - if (c) - *c = '\0'; - if (!strcasecmp(cmd, "SIP/2.0")) + /* In principle Call-ID's uniquely identify a call, but with a forking SIP proxy + we need more to identify a branch - so we have to check branch, from + and to tags to identify a call leg. + For Asterisk to behave correctly, you need to turn on pedanticsipchecking + in sip.conf + */ + if (req->method == SIP_RESPONSE) ast_copy_string(tmp, get_header(req, "To"), sizeof(tmp)); else ast_copy_string(tmp, get_header(req, "From"), sizeof(tmp)); @@ -2911,15 +2914,16 @@ } - if (ast_strlen_zero(callid)) { - ast_log(LOG_WARNING, "Call missing call ID from '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin->sin_addr)); - return NULL; - } ast_mutex_lock(&iflock); p = iflist; while(p) { - if (!strcmp(p->callid, callid) && - (!pedanticsipchecking || !tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) { + int found = 0; + if (req->method == SIP_REGISTER) + found = (!strcmp(p->callid, callid)); + else + found = (!strcmp(p->callid, callid) && + (!pedanticsipchecking || !tag || ast_strlen_zero(p->theirtag) || !strcmp(p->theirtag, tag))) ; + if (found) { /* Found the call */ ast_mutex_lock(&p->lock); ast_mutex_unlock(&iflock); @@ -3053,12 +3057,13 @@ return t; } -/*--- parse: Parse a SIP message ----*/ -static void parse(struct sip_request *req) +/*--- parse_request: Parse a SIP message ----*/ +static void parse_request(struct sip_request *req) { /* Divide fields by NULL's */ char *c; int f = 0; + c = req->data; /* First header starts immediately */ @@ -3068,16 +3073,15 @@ /* We've got a new header */ *c = 0; -#if 0 - printf("Header: %s (%d)\n", req->header[f], strlen(req->header[f])); -#endif + if (option_debug > 3) + ast_log(LOG_DEBUG, "Header: %s (%d)\n", req->header[f], strlen(req->header[f])); if (ast_strlen_zero(req->header[f])) { /* Line by itself means we're now in content */ c++; break; } if (f >= SIP_MAX_HEADERS - 1) { - ast_log(LOG_WARNING, "Too many SIP headers...\n"); + ast_log(LOG_WARNING, "Too many SIP headers. Ignoring.\n"); } else f++; req->header[f] = c + 1; @@ -3098,11 +3102,10 @@ if (*c == '\n') { /* We've got a new line */ *c = 0; -#if 0 - printf("Line: %s (%d)\n", req->line[f], strlen(req->line[f])); -#endif + if (option_debug > 3) + ast_log(LOG_DEBUG, "Line: %s (%d)\n", req->line[f], strlen(req->line[f])); if (f >= SIP_MAX_LINES - 1) { - ast_log(LOG_WARNING, "Too many SDP lines...\n"); + ast_log(LOG_WARNING, "Too many SDP lines. Ignoring.\n"); } else f++; req->line[f] = c + 1; @@ -3118,6 +3121,8 @@ req->lines = f; if (*c) ast_log(LOG_WARNING, "Odd content, extra stuff left over ('%s')\n", c); + /* Split up the first line parts */ + determine_firstline_parts(req); } /*--- process_sdp: Process SIP SDP and activate RTP channels---*/ @@ -3637,6 +3642,7 @@ ast_log(LOG_WARNING, "Request already initialized?!?\n"); return -1; } + req->method = SIP_RESPONSE; req->header[req->headers] = req->data + req->len; snprintf(req->header[req->headers], sizeof(req->data) - req->len, "SIP/2.0 %s\r\n", resp); req->len += strlen(req->header[req->headers]); @@ -4208,10 +4214,9 @@ add_sdp(&req, p); /* Use this as the basis */ copy_request(&p->initreq, &req); - parse(&p->initreq); + parse_request(&p->initreq); if (sip_debug_test_pvt(p)) ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - determine_firstline_parts(&p->initreq); p->lastinvite = p->ocseq; ast_set_flag(p, SIP_OUTGOING); return send_request(p, &req, 1, p->ocseq); @@ -4358,6 +4363,7 @@ { struct sip_request req; + req.method = sipmethod; if (init) { /* Bump branch even on initial requests */ p->branch ^= rand(); @@ -4440,10 +4446,9 @@ if (!p->initreq.headers) { /* Use this as the basis */ copy_request(&p->initreq, &req); - parse(&p->initreq); + parse_request(&p->initreq); if (sip_debug_test_pvt(p)) ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); - determine_firstline_parts(&p->initreq); } p->lastinvite = p->ocseq; return send_request(p, &req, init ? 2 : 1, p->ocseq); @@ -4584,7 +4589,7 @@ if (!p->initreq.headers) { /* Use this as the basis */ copy_request(&p->initreq, &req); - parse(&p->initreq); + parse_request(&p->initreq); if (sip_debug_test_pvt(p)) ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); determine_firstline_parts(&p->initreq); @@ -4599,7 +4604,7 @@ if (!p->initreq.headers) { /* Use this as the basis */ copy_request(&p->initreq, req); - parse(&p->initreq); + parse_request(&p->initreq); if (sip_debug_test_pvt(p)) ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); determine_firstline_parts(&p->initreq); @@ -4632,7 +4637,7 @@ if (!p->initreq.headers) { /* Use this as the basis */ copy_request(&p->initreq, &req); - parse(&p->initreq); + parse_request(&p->initreq); if (sip_debug_test_pvt(p)) ast_verbose("%d headers, %d lines\n", p->initreq.headers, p->initreq.lines); determine_firstline_parts(&p->initreq); @@ -4908,7 +4913,7 @@ add_header(&req, "Content-Length", "0"); add_blank_header(&req); copy_request(&p->initreq, &req); - parse(&p->initreq); + parse_request(&p->initreq); if (sip_debug_test_pvt(p)) { ast_verbose("REGISTER %d headers, %d lines\n", p->initreq.headers, p->initreq.lines); } @@ -9037,7 +9042,7 @@ } } } else if (debug) - ast_verbose("Ignoring this request\n"); + ast_verbose("Ignoring this INVITE request\n"); if (!p->lastinvite && !ignore && !p->owner) { /* Handle authentication if this is our first invite */ res = check_user(p, req, SIP_INVITE, e, 1, sin, ignore); @@ -9563,7 +9568,8 @@ /* Save useragent of the client */ useragent = get_header(req, "User-Agent"); - ast_copy_string(p->useragent, useragent, sizeof(p->useragent)); + if (!ast_strlen_zero(useragent)) + ast_copy_string(p->useragent, useragent, sizeof(p->useragent)); /* Find out SIP method for incoming request */ if (!strcasecmp(cmd, "SIP/2.0")) { /* Response to our request */ @@ -9595,18 +9601,20 @@ /* New SIP request coming in (could be new request in existing SIP dialog as well...) */ - p->method = find_sip_method(cmd); /* Find out which SIP method they are using */ + p->method = req->method; /* Find out which SIP method they are using */ if (option_debug > 2) ast_log(LOG_DEBUG, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd); if (p->icseq && (p->icseq > seqno)) { ast_log(LOG_DEBUG, "Ignoring too old SIP packet packet %d (expecting >= %d)\n", seqno, p->icseq); return -1; - } else if (p->icseq && (p->icseq == seqno) && (strcasecmp(cmd, "CANCEL") || ast_test_flag(p, SIP_ALREADYGONE))) { + } else if (p->icseq && (p->icseq == seqno) && (p->method != SIP_CANCEL|| ast_test_flag(p, SIP_ALREADYGONE))) { /* ignore means "don't do anything with it" but still have to respond appropriately. We do this if we receive a repeat of the last sequence number */ - ignore=1; + ignore=2; + if (option_debug > 2) + ast_log(LOG_DEBUG, "Ignoring SIP message because of retransmit (%s Seqno %d, ours %d)\n", sip_methods[p->method].text, p->icseq, seqno); } if (seqno >= p->icseq) @@ -9708,7 +9716,6 @@ socklen_t len; int nounlock; int recount = 0; - int debug; char iabuf[INET_ADDRSTRLEN]; len = sizeof(sin); @@ -9726,13 +9733,14 @@ } req.data[res] = '\0'; req.len = res; - debug = sip_debug_test_addr(&sin); + req.debug = sip_debug_test_addr(&sin); if (pedanticsipchecking) - req.len = lws2sws(req.data, req.len); - if (debug) + req.len = lws2sws(req.data, req.len); /* Fix multiline headers */ + if (req.debug) ast_verbose("\n<-- SIP read from %s:%d: \n%s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), sin.sin_addr), ntohs(sin.sin_port), req.data); - parse(&req); - if (debug) { + parse_request(&req); + req.method = find_sip_method(req.rlPart1); + if (req.debug) { ast_verbose("--- (%d headers %d lines)", req.headers, req.lines); if (req.headers + req.lines == 0) ast_verbose(" Nat keepalive "); @@ -9744,9 +9752,6 @@ return 1; } - /* Determine the request URI for sip, sips or tel URIs */ - if (determine_firstline_parts(&req) < 0) - return 1; /* Process request, with netlock held */ retrylock: @@ -10111,7 +10116,7 @@ /*--- sip_request: PBX interface function -build SIP pvt structure ---*/ /* SIP calls initiated by the PBX arrive here */ -static struct ast_channel *sip_request(const char *type, int format, void *data, int *cause) +static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause) { int oldformat; struct sip_pvt *p;