chan_sip.c

	return "";
}

/*! \brief Get a line from an SDP message body */
static const char *get_sdp(struct sip_request *req, const char *name) 
{
	int dummy = 0;

	return get_sdp_iterate(&dummy, req, name);
}

/*! \brief Get a specific line from the message body */
static char *get_body(struct sip_request *req, char *name) 
{
	int x;
	int len = strlen(name);
	char *r;

	for (x = 0; x < req->lines; x++) {
		r = get_body_by_line(req->line[x], name, len);
		if (r[0] != '\0')
			return r;
	}

	return "";
}

/*! \brief Find compressed SIP alias */
static const char *find_alias(const char *name, const char *_default)
{
	/*! \brief Structure for conversion between compressed SIP and "normal" SIP */
	static const struct cfalias {
		char * const fullname;
		char * const shortname;
	} aliases[] = {
		{ "Content-Type",	 "c" },
		{ "Content-Encoding",	 "e" },
		{ "From",		 "f" },
		{ "Call-ID",		 "i" },
		{ "Contact",		 "m" },
		{ "Content-Length",	 "l" },
		{ "Subject",		 "s" },
		{ "To",			 "t" },
		{ "Supported",		 "k" },
		{ "Refer-To",		 "r" },
		{ "Referred-By",	 "b" },
		{ "Allow-Events",	 "u" },
		{ "Event",		 "o" },
		{ "Via",		 "v" },
		{ "Accept-Contact",      "a" },
		{ "Reject-Contact",      "j" },
		{ "Request-Disposition", "d" },
		{ "Session-Expires",     "x" },
		{ "Identity",            "y" },
		{ "Identity-Info",       "n" },
	};
	int x;

	for (x=0; x<sizeof(aliases) / sizeof(aliases[0]); x++) 
		if (!strcasecmp(aliases[x].fullname, name))
			return aliases[x].shortname;

	return _default;
}

static const char *__get_header(const struct sip_request *req, const char *name, int *start)
{
	int pass;

	/*
	 * Technically you can place arbitrary whitespace both before and after the ':' in
	 * a header, although RFC3261 clearly says you shouldn't before, and place just
	 * one afterwards.  If you shouldn't do it, what absolute idiot decided it was 
	 * a good idea to say you can do it, and if you can do it, why in the hell would.
	 * you say you shouldn't.
	 * Anyways, pedanticsipchecking controls whether we allow spaces before ':',
	 * and we always allow spaces after that for compatibility.
	 */
	for (pass = 0; name && pass < 2;pass++) {
		int x, len = strlen(name);
		for (x=*start; x<req->headers; x++) {
			if (!strncasecmp(req->header[x], name, len)) {
				char *r = req->header[x] + len;	/* skip name */
				if (pedanticsipchecking)
					r = ast_skip_blanks(r);

				if (*r == ':') {
					*start = x+1;
					return ast_skip_blanks(r+1);
				}
			}
		}
		if (pass == 0) /* Try aliases */
			name = find_alias(name, NULL);
	}

	/* Don't return NULL, so get_header is always a valid pointer */
	return "";
}

/*! \brief Get header from SIP request 
	\return Always return something, so don't check for NULL because it won't happen :-)
*/
static const char *get_header(const struct sip_request *req, const char *name)
{
	int start = 0;
	return __get_header(req, name, &start);
}

/*! \brief Read RTP from network */
static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p, int *faxdetect)
{
	/* Retrieve audio/etc from channel.  Assumes p->lock is already held. */
	struct ast_frame *f;
	
	if (!p->rtp) {
		/* We have no RTP allocated for this channel */
		return &ast_null_frame;
	}

	switch(ast->fdno) {
	case 0:
		f = ast_rtp_read(p->rtp);	/* RTP Audio */
		break;
	case 1:
		f = ast_rtcp_read(p->rtp);	/* RTCP Control Channel */
		break;
	case 2:
		f = ast_rtp_read(p->vrtp);	/* RTP Video */
		break;
	case 3:
		f = ast_rtcp_read(p->vrtp);	/* RTCP Control Channel for video */
		break;
	case 5:
		f = ast_udptl_read(p->udptl);	/* UDPTL for T.38 */
		break;
	default:
		f = &ast_null_frame;
	}
	/* Don't forward RFC2833 if we're not supposed to */
	if (f && (f->frametype == AST_FRAME_DTMF) &&
	    (ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_RFC2833))
		return &ast_null_frame;

	if (p->owner) {
		/* We already hold the channel lock */
		if (f->frametype == AST_FRAME_VOICE) {
			if (f->subclass != (p->owner->nativeformats & AST_FORMAT_AUDIO_MASK)) {
				if (option_debug)
					ast_log(LOG_DEBUG, "Oooh, format changed to %d\n", f->subclass);
				p->owner->nativeformats = (p->owner->nativeformats & AST_FORMAT_VIDEO_MASK) | f->subclass;
				ast_set_read_format(p->owner, p->owner->readformat);
				ast_set_write_format(p->owner, p->owner->writeformat);
			}
			if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_INBAND) && p->vad) {
				f = ast_dsp_process(p->owner, p->vad, f);
				if (f && f->frametype == AST_FRAME_DTMF) {
					if (ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && f->subclass == 'f') {
						if (option_debug)
							ast_log(LOG_DEBUG, "Fax CNG detected on %s\n", ast->name);
						*faxdetect = 1;
					} else if (option_debug) {
						ast_log(LOG_DEBUG, "* Detected inband DTMF '%c'\n", f->subclass);
					}
				}
			}
		}
	}
	return f;
}

/*! \brief Read SIP RTP from channel */
static struct ast_frame *sip_read(struct ast_channel *ast)
{
	struct ast_frame *fr;
	struct sip_pvt *p = ast->tech_pvt;
	int faxdetected = FALSE;

	sip_pvt_lock(p);
	fr = sip_rtp_read(ast, p, &faxdetected);
	p->lastrtprx = time(NULL);

	/* If we are NOT bridged to another channel, and we have detected fax tone we issue T38 re-invite to a peer */
	/* If we are bridged then it is the responsibility of the SIP device to issue T38 re-invite if it detects CNG or fax preamble */
	if (faxdetected && ast_test_flag(&p->t38.t38support, SIP_PAGE2_T38SUPPORT_UDPTL) && (p->t38.state == T38_DISABLED) && !(ast_bridged_channel(ast))) {
		if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
			if (!p->pendinginvite) {
				if (option_debug > 2)
					ast_log(LOG_DEBUG, "Sending reinvite on SIP (%s) for T.38 negotiation.\n",ast->name);
				p->t38.state = T38_LOCAL_REINVITE;
				transmit_reinvite_with_sdp(p, TRUE);
				if (option_debug > 1)
					ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, ast->name);
			}
		} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
			if (option_debug > 2)
				ast_log(LOG_DEBUG, "Deferring reinvite on SIP (%s) - it will be re-negotiated for T.38\n", ast->name);
			ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
		}
	}

	sip_pvt_unlock(p);
	return fr;
}


/*! \brief Generate 32 byte random string for callid's etc */
static char *generate_random_string(char *buf, size_t size)
{
	long val[4];
	int x;

	for (x=0; x<4; x++)
		val[x] = ast_random();
	snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]);

	return buf;
}

/*! \brief Build SIP Call-ID value for a non-REGISTER transaction */
static void build_callid_pvt(struct sip_pvt *pvt)
{
	char buf[33];

	const char *host = S_OR(pvt->fromdomain, ast_inet_ntoa(pvt->ourip));
	
	ast_string_field_build(pvt, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);

}

/*! \brief Build SIP Call-ID value for a REGISTER transaction */
static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain)
{
	char buf[33];

	const char *host = S_OR(fromdomain, ast_inet_ntoa(ourip));

	ast_string_field_build(reg, callid, "%s@%s", generate_random_string(buf, sizeof(buf)), host);
}

/*! \brief Make our SIP dialog tag */
static void make_our_tag(char *tagbuf, size_t len)
{
	snprintf(tagbuf, len, "as%08lx", ast_random());
}

/*! \brief Allocate SIP_PVT structure and set defaults */
static struct sip_pvt *sip_alloc(ast_string_field callid, struct sockaddr_in *sin,
				 int useglobal_nat, const int intended_method)
{
	struct sip_pvt *p;

	if (!(p = ast_calloc(1, sizeof(*p))))
		return NULL;

	if (ast_string_field_init(p, 512)) {
		free(p);
		return NULL;
	}

	ast_mutex_init(&p->pvt_lock);

	p->method = intended_method;
	p->initid = -1;
	p->autokillid = -1;
	p->subscribed = NONE;
	p->stateid = -1;
	p->prefs = default_prefs;		/* Set default codecs for this call */

	if (intended_method != SIP_OPTIONS)	/* Peerpoke has it's own system */
		p->timer_t1 = SIP_TIMER_T1;	/* Default SIP retransmission timer T1 (RFC 3261) */

	if (sin) {
		p->sa = *sin;
		if (ast_sip_ouraddrfor(&p->sa.sin_addr, &p->ourip))
			p->ourip = __ourip;
	} else
		p->ourip = __ourip;

	/* Copy global flags to this PVT at setup. */
	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;

	if (sip_methods[intended_method].need_rtp) {
		p->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
		/* If the global videosupport flag is on, we always create a RTP interface for video */
		if (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT))
			p->vrtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
		if (ast_test_flag(&p->flags[1], SIP_PAGE2_T38SUPPORT))
			p->udptl = ast_udptl_new_with_bindaddr(sched, io, 0, bindaddr.sin_addr);
		if (!p->rtp || (ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && !p->vrtp)) {
			ast_log(LOG_WARNING, "Unable to create RTP audio %s session: %s\n",
				ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) ? "and video" : "", strerror(errno));
			ast_mutex_destroy(&p->pvt_lock);
			if (p->chanvars) {
				ast_variables_destroy(p->chanvars);
				p->chanvars = NULL;
			}
			free(p);
			return NULL;
		}
		ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) != SIP_DTMF_INFO);
		ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
		ast_rtp_settos(p->rtp, global_tos_audio);
		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_settos(p->vrtp, global_tos_video);
			ast_rtp_setdtmf(p->vrtp, 0);
			ast_rtp_setdtmfcompensate(p->vrtp, 0);
			ast_rtp_set_rtptimeout(p->vrtp, global_rtptimeout);
			ast_rtp_set_rtpholdtimeout(p->vrtp, global_rtpholdtimeout);
			ast_rtp_set_rtpkeepalive(p->vrtp, global_rtpkeepalive);
		}
		if (p->udptl)
			ast_udptl_settos(p->udptl, global_tos_audio);
		p->maxcallbitrate = default_maxcallbitrate;
	}

	if (useglobal_nat && sin) {
		/* Setup NAT structure according to global settings if we have an address */
		ast_copy_flags(&p->flags[0], &global_flags[0], SIP_NAT);
		p->recv = *sin;
		do_setnat(p, ast_test_flag(&p->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
	}

	if (p->method != SIP_REGISTER)
		ast_string_field_set(p, fromdomain, default_fromdomain);
	build_via(p);
	if (!callid)
		build_callid_pvt(p);
	else
		ast_string_field_set(p, callid, callid);
	/* Assign default music on hold class */
	ast_string_field_set(p, mohinterpret, default_mohinterpret);
	ast_string_field_set(p, mohsuggest, default_mohsuggest);
	p->capability = global_capability;
	p->allowtransfer = global_allowtransfer;
	if ((ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
	    (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
		p->noncodeccapability |= AST_RTP_DTMF;
	if (p->udptl) {
		p->t38.capability = global_t38_capability;
		if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_REDUNDANCY)
			p->t38.capability |= T38FAX_UDP_EC_REDUNDANCY;
		else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_FEC)
			p->t38.capability |= T38FAX_UDP_EC_FEC;
		else if (ast_udptl_get_error_correction_scheme(p->udptl) == UDPTL_ERROR_CORRECTION_NONE)
			p->t38.capability |= T38FAX_UDP_EC_NONE;
		p->t38.capability |= T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF;
		p->t38.jointcapability = p->t38.capability;
	}
	ast_string_field_set(p, context, default_context);

	/* Add to active dialog list */
	dialoglist_lock();
	p->next = dialoglist;
	dialoglist = p;
	dialoglist_unlock();
	if (option_debug)
		ast_log(LOG_DEBUG, "Allocating new SIP dialog for %s - %s (%s)\n", callid ? callid : "(No Call-ID)", sip_methods[intended_method].text, p->rtp ? "With RTP" : "No RTP");
	return p;
}

/*! \brief Connect incoming SIP message to current dialog or create new dialog structure
	Called by handle_request, sipsock_read */
static struct sip_pvt *find_call(struct sip_request *req, struct sockaddr_in *sin, const int intended_method)
{
	struct sip_pvt *p = NULL;
	char *tag = "";	/* note, tag is never NULL */
	char totag[128];
	char fromtag[128];
	const char *callid = get_header(req, "Call-ID");
	const char *from = get_header(req, "From");
	const char *to = get_header(req, "To");
	const char *cseq = get_header(req, "Cseq");

	/* Call-ID, to, from and Cseq are required by RFC 3261. (Max-forwards and via too - ignored now) */
	/* get_header always returns non-NULL so we must use ast_strlen_zero() */
	if (ast_strlen_zero(callid) || ast_strlen_zero(to) ||
			ast_strlen_zero(from) || ast_strlen_zero(cseq))
		return NULL;	/* Invalid packet */

	if (pedanticsipchecking) {
		/* 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 (gettag(req, "To", totag, sizeof(totag)))
			ast_set_flag(req, SIP_PKT_WITH_TOTAG);	/* Used in handle_request/response */
		gettag(req, "From", fromtag, sizeof(fromtag));

		tag = (req->method == SIP_RESPONSE) ? totag : fromtag;

		if (option_debug > 4 )
			ast_log(LOG_DEBUG, "= Looking for  Call ID: %s (Checking %s) --From tag %s --To-tag %s  \n", callid, req->method==SIP_RESPONSE ? "To" : "From", fromtag, totag);

		/* All messages must always have From: tag */
		if (ast_strlen_zero(fromtag)) {
			if (option_debug > 4 ) 
				ast_log(LOG_DEBUG, "%s request has no from tag, dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
			return NULL;
		}
		/* reject requests that must always have a To: tag */
		if (ast_strlen_zero(totag) && (req->method == SIP_ACK || req->method == SIP_BYE || req->method == SIP_INFO )) {
			if (option_debug > 4) 
				ast_log(LOG_DEBUG, "%s must have a to tag. dropping callid: %s from: %s\n", sip_methods[req->method].text , callid, from );
			return NULL;
		}
	}

	dialoglist_lock();
	for (p = dialoglist; p; p = p->next) {
		/* In pedantic, we do not want packets with bad syntax to be connected to a PVT */
		int found = FALSE;
		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 (option_debug > 4)
			ast_log(LOG_DEBUG, "= %s Their Call ID: %s Their Tag %s Our tag: %s\n", found ? "Found" : "No match", p->callid, p->theirtag, p->tag);

		/* If we get a new request within an existing to-tag - check the to tag as well */
		if (pedanticsipchecking && found  && req->method != SIP_RESPONSE) {	/* SIP Request */
			if (p->tag[0] == '\0' && totag[0]) {
				/* We have no to tag, but they have. Wrong dialog */
				found = FALSE;
			} else if (totag[0]) {			/* Both have tags, compare them */
				if (strcmp(totag, p->tag)) {
					found = FALSE;		/* This is not our packet */
				}
			}
			if (!found && option_debug > 4)
				ast_log(LOG_DEBUG, "= Being pedantic: This is not our match on request: Call ID: %s Ourtag <null> Totag %s Method %s\n", p->callid, totag, sip_methods[req->method].text);
		}


		if (found) {
			/* Found the call */
			sip_pvt_lock(p);
			dialoglist_unlock();
			return p;
		}
	}
	dialoglist_unlock();
	
	/* See if the method is capable of creating a dialog */
	if (sip_methods[intended_method].can_create == CAN_CREATE_DIALOG) {
		if (intended_method == SIP_REFER) {
			/* We do support REFER, but not outside of a dialog yet */
			transmit_response_using_temp(callid, sin, 1, intended_method, req, "603 Declined (no dialog)");
		} else if (intended_method == SIP_NOTIFY) {
			/* We do not support out-of-dialog NOTIFY either,
		   	like voicemail notification, so cancel that early */
			transmit_response_using_temp(callid, sin, 1, intended_method, req, "489 Bad event");
		} else {
			/* Ok, time to create a new SIP dialog object, a pvt */
			if ((p = sip_alloc(callid, sin, 1, intended_method)))  {
				/* Ok, we've created a dialog, let's go and process it */
				sip_pvt_lock(p);
			} else {
				/* We have a memory or file/socket error (can't allocate RTP sockets or something) so we're not
					getting a dialog from sip_alloc. 
	
					Without a dialog we can't retransmit and handle ACKs and all that, but at least
					send an error message.
	
					Sorry, we apologize for the inconvienience
				*/
				transmit_response_using_temp(callid, sin, 1, intended_method, req, "500 Server internal error");
				if (option_debug > 3)
					ast_log(LOG_DEBUG, "Failed allocating SIP dialog, sending 500 Server internal error and giving up\n");
			}
		}
		return p;
	} else if( sip_methods[intended_method].can_create == CAN_CREATE_DIALOG_UNSUPPORTED_METHOD) {
		/* A method we do not support, let's take it on the volley */
		transmit_response_using_temp(callid, sin, 1, intended_method, req, "501 Method Not Implemented");
		if (option_debug > 1 )
			ast_log(LOG_DEBUG, "Got a request with unsupported SIP method.\n");
	} else if (intended_method != SIP_RESPONSE && intended_method != SIP_ACK) {
		/* This is a request outside of a dialog that we don't know about */
		transmit_response_using_temp(callid, sin, 1, intended_method, req, "481 Call leg/transaction does not exist");
		if (option_debug > 1)
			ast_log(LOG_DEBUG, "That's odd...  Got a request in unknown dialog. Callid %s\n", callid ? callid : "<unknown>");
	}
	/* We do not respond to responses for dialogs that we don't know about, we just drop
	   the session quickly */
	if (option_debug > 1 && intended_method == SIP_RESPONSE)
		ast_log(LOG_DEBUG, "That's odd...  Got a response on a call we dont know about. Callid %s\n", callid ? callid : "<unknown>");

	return p;
}

/*! \brief Parse register=> line in sip.conf and add to registry */
static int sip_register(char *value, int lineno)
{
	struct sip_registry *reg;
	int portnum = 0;
	char username[256] = "";
	char *hostname=NULL, *secret=NULL, *authuser=NULL;
	char *porta=NULL;
	char *callback=NULL;

	if (!value)
		return -1;
	ast_copy_string(username, value, sizeof(username));
	/* First split around the last '@' then parse the two components. */
	hostname = strrchr(username, '@'); /* allow @ in the first part */
	if (hostname)
		*hostname++ = '\0';
	if (ast_strlen_zero(username) || ast_strlen_zero(hostname)) {
		ast_log(LOG_WARNING, "Format for registration is user[:secret[:authuser]]@host[:port][/contact] at line %d\n", lineno);
		return -1;
	}
	/* split user[:secret[:authuser]] */
	secret = strchr(username, ':');
	if (secret) {
		*secret++ = '\0';
		authuser = strchr(secret, ':');
		if (authuser)
			*authuser++ = '\0';
	}
	/* split host[:port][/contact] */
	callback = strchr(hostname, '/');
	if (callback)
		*callback++ = '\0';
	if (ast_strlen_zero(callback))
		callback = "s";
	porta = strchr(hostname, ':');
	if (porta) {
		*porta++ = '\0';
		portnum = atoi(porta);
		if (portnum == 0) {
			ast_log(LOG_WARNING, "%s is not a valid port number at line %d\n", porta, lineno);
			return -1;
		}
	}
	if (!(reg = ast_calloc(1, sizeof(*reg)))) {
		ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry entry\n");
		return -1;
	}

	if (ast_string_field_init(reg, 256)) {
		ast_log(LOG_ERROR, "Out of memory. Can't allocate SIP registry strings\n");
		free(reg);
		return -1;
	}

	regobjs++;
	ASTOBJ_INIT(reg);
	ast_string_field_set(reg, callback, callback);
	if (username)
		ast_string_field_set(reg, username, username);
	if (hostname)
		ast_string_field_set(reg, hostname, hostname);
	if (authuser)
		ast_string_field_set(reg, authuser, authuser);
	if (secret)
		ast_string_field_set(reg, secret, secret);
	reg->expire = -1;
	reg->expiry = default_expiry;
	reg->timeout =  -1;
	reg->refresh = default_expiry;
	reg->portno = portnum;
	reg->callid_valid = FALSE;
	reg->ocseq = INITIAL_CSEQ;
	ASTOBJ_CONTAINER_LINK(&regl, reg);	/* Add the new registry entry to the list */
	unref_registry(reg);
	return 0;
}

/*! \brief  Parse multiline SIP headers into one header
	This is enabled if pedanticsipchecking is enabled */
static int lws2sws(char *msgbuf, int len) 
{
	int h = 0, t = 0; 
	int lws = 0; 

	for (; h < len;) { 
		/* Eliminate all CRs */ 
		if (msgbuf[h] == '\r') { 
			h++; 
			continue; 
		} 
		/* Check for end-of-line */ 
		if (msgbuf[h] == '\n') { 
			/* Check for end-of-message */ 
			if (h + 1 == len) 
				break; 
			/* Check for a continuation line */ 
			if (msgbuf[h + 1] == ' ' || msgbuf[h + 1] == '\t') { 
				/* Merge continuation line */ 
				h++; 
				continue; 
			} 
			/* Propagate LF and start new line */ 
			msgbuf[t++] = msgbuf[h++]; 
			lws = 0;
			continue; 
		} 
		if (msgbuf[h] == ' ' || msgbuf[h] == '\t') { 
			if (lws) { 
				h++; 
				continue; 
			} 
			msgbuf[t++] = msgbuf[h++]; 
			lws = 1; 
			continue; 
		} 
		msgbuf[t++] = msgbuf[h++]; 
		if (lws) 
			lws = 0; 
	} 
	msgbuf[t] = '\0'; 
	return t; 
}

/*! \brief Parse a SIP message 
	\note this function is used both on incoming and outgoing packets
*/
static void parse_request(struct sip_request *req)
{
	char *c = req->data, **dst = req->header;
	int i = 0, lim = SIP_MAX_HEADERS - 1;

	req->header[0] = c;
	req->headers = -1;	/* mark that we are working on the header */
	for (; *c; c++) {
		if (*c == '\r')		/* remove \r */
			*c = '\0';
		else if (*c == '\n') { /* end of this line */
			*c = '\0';
			if (sipdebug && option_debug > 3)
				ast_log(LOG_DEBUG, "%7s %2d [%3d]: %s\n",
					req->headers < 0 ? "Header" : "Body",
					i, (int)strlen(dst[i]), dst[i]);
			if (ast_strlen_zero(dst[i]) && req->headers < 0) {
				req->headers = i;	/* record number of header lines */
				dst = req->line;	/* start working on the body */
				i = 0;
				lim = SIP_MAX_LINES - 1;
			} else {	/* move to next line, check for overflows */
				if (i++ >= lim)
					break;
			}
			dst[i] = c + 1; /* record start of next line */
		}
        }
	/* update count of header or body lines */
	if (req->headers >= 0)	/* we are in the body */
		req->lines = i;
	else {			/* no body */
		req->headers = i;
		req->lines = 0;
		req->line[0] = "";
	}

	if (*c)
		ast_log(LOG_WARNING, "Too many lines, skipping <%s>\n", c);
	/* Split up the first line parts */
	determine_firstline_parts(req);
}

/*!
  \brief Determine whether a SIP message contains an SDP in its body
  \param req the SIP request to process
  \return 1 if SDP found, 0 if not found

  Also updates req->sdp_start and req->sdp_end to indicate where the SDP
  lives in the message body.
*/
static int find_sdp(struct sip_request *req)
{
	const char *content_type;
	const char *search;
	char *boundary;
	unsigned int x;

	content_type = get_header(req, "Content-Type");

	/* if the body contains only SDP, this is easy */
	if (!strcasecmp(content_type, "application/sdp")) {
		req->sdp_start = 0;
		req->sdp_end = req->lines;
		return 1;
	}

	/* if it's not multipart/mixed, there cannot be an SDP */
	if (strncasecmp(content_type, "multipart/mixed", 15))
		return 0;

	/* if there is no boundary marker, it's invalid */
	if (!(search = strcasestr(content_type, ";boundary=")))
		return 0;

	search += 10;

	if (ast_strlen_zero(search))
		return 0;

	/* make a duplicate of the string, with two extra characters
	   at the beginning */
	boundary = ast_strdupa(search - 2);
	boundary[0] = boundary[1] = '-';

	/* search for the boundary marker, but stop when there are not enough
	   lines left for it, the Content-Type header and at least one line of
	   body */
	for (x = 0; x < (req->lines - 2); x++) {
		if (!strncasecmp(req->line[x], boundary, strlen(boundary)) &&
		    !strcasecmp(req->line[x + 1], "Content-Type: application/sdp")) {
			x += 2;
			req->sdp_start = x;

			/* search for the end of the body part */
			for ( ; x < req->lines; x++) {
				if (!strncasecmp(req->line[x], boundary, strlen(boundary)))
					break;
			}
			req->sdp_end = x;
			return 1;
		}
	}

	return 0;
}

/*! \brief Process SIP SDP offer, select formats and activate RTP channels
	If offer is rejected, we will not change any properties of the call
 	Return 0 on success, a negative value on errors.
	Must be called after find_sdp().
*/
static int process_sdp(struct sip_pvt *p, struct sip_request *req)
{
	const char *m;		/* SDP media offer */
	const char *c;
	const char *a;
	char host[258];
	int len = -1;
	int portno = -1;		/*!< RTP Audio port number */
	int vportno = -1;		/*!< RTP Video port number */
	int udptlportno = -1;
	int peert38capability = 0;
	char s[256];
	int old = 0;

	/* Peer capability is the capability in the SDP, non codec is RFC2833 DTMF (101) */	
	int peercapability = 0, peernoncodeccapability = 0;
	int vpeercapability = 0, vpeernoncodeccapability = 0;
	struct sockaddr_in sin;		/*!< media socket address */
	struct sockaddr_in vsin;	/*!< Video socket address */

	const char *codecs;
	struct hostent *hp;		/*!< RTP Audio host IP */
	struct hostent *vhp = NULL;	/*!< RTP video host IP */
	struct ast_hostent audiohp;
	struct ast_hostent videohp;
	int codec;
	int destiterator = 0;
	int iterator;
	int sendonly = 0;
	int numberofports;
	struct ast_channel *bridgepeer = NULL;
	struct ast_rtp *newaudiortp, *newvideortp;	/* Buffers for codec handling */
	int newjointcapability;				/* Negotiated capability */
	int newpeercapability;
	int newnoncodeccapability;
	int numberofmediastreams = 0;
	int debug = sip_debug_test_pvt(p);
		
	int found_rtpmap_codecs[32];
	int last_rtpmap_codec=0;

	if (!p->rtp) {
		ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
		return -1;
	}

	/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
	newaudiortp = alloca(ast_rtp_alloc_size());
	memset(newaudiortp, 0, ast_rtp_alloc_size());
	ast_rtp_new_init(newaudiortp);
	ast_rtp_pt_clear(newaudiortp);

	newvideortp = alloca(ast_rtp_alloc_size());
	memset(newvideortp, 0, ast_rtp_alloc_size());
	ast_rtp_new_init(newvideortp);
	ast_rtp_pt_clear(newvideortp);

	/* Update our last rtprx when we receive an SDP, too */
	p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */


	/* Try to find first media stream */
	m = get_sdp(req, "m");
	destiterator = req->sdp_start;
	c = get_sdp_iterate(&destiterator, req, "c");
	if (ast_strlen_zero(m) || ast_strlen_zero(c)) {
		ast_log(LOG_WARNING, "Insufficient information for SDP (m = '%s', c = '%s')\n", m, c);
		return -1;
	}

	/* Check for IPv4 address (not IPv6 yet) */
	if (sscanf(c, "IN IP4 %256s", host) != 1) {
		ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
		return -1;
	}

	/* XXX This could block for a long time, and block the main thread! XXX */
	hp = ast_gethostbyname(host, &audiohp);
	if (!hp) {
		ast_log(LOG_WARNING, "Unable to lookup host in c= line, '%s'\n", c);
		return -1;
	}
	vhp = hp;	/* Copy to video address as default too */
	
	iterator = req->sdp_start;
	ast_set_flag(&p->flags[0], SIP_NOVIDEO);	


	/* Find media streams in this SDP offer */
	while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
		int x;
		int audio = FALSE;

		numberofports = 1;
		if ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
		    (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1)) {
			audio = TRUE;
			numberofmediastreams++;
			/* Found audio stream in this media definition */
			portno = x;
			/* Scan through the RTP payload types specified in a "m=" line: */
			for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
				if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
					ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
					return -1;
				}
				if (debug)
					ast_verbose("Found RTP audio format %d\n", codec);
				ast_rtp_set_m_type(newaudiortp, codec);
			}
		} else if ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2) ||
		    (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1)) {
			/* If it is not audio - is it video ? */
			ast_clear_flag(&p->flags[0], SIP_NOVIDEO);	
			numberofmediastreams++;
			vportno = x;
			/* Scan through the RTP payload types specified in a "m=" line: */
			for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
				if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
					ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
					return -1;
				}
				if (debug)
					ast_verbose("Found RTP video format %d\n", codec);
				ast_rtp_set_m_type(newvideortp, codec);
			}
		} else if (p->udptl && ( (sscanf(m, "image %d udptl t38%n", &x, &len) == 1) || 
		 (sscanf(m, "image %d UDPTL t38%n", &x, &len) == 1) )) {
			if (debug)
				ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
			udptlportno = x;
			numberofmediastreams++;
			
			if (p->owner && p->lastinvite) {
				p->t38.state = T38_PEER_REINVITE; /* T38 Offered in re-invite from remote party */
				if (option_debug > 1)
					ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>" );
			} else {
				p->t38.state = T38_PEER_DIRECT; /* T38 Offered directly from peer in first invite */
				if (option_debug > 1)
					ast_log(LOG_DEBUG, "T38 state changed to %d on channel %s\n", p->t38.state, p->owner ? p->owner->name : "<none>");
			}
		} else 
			ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
		if (numberofports > 1)
			ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
		

		/* Check for Media-description-level-address for audio */
		c = get_sdp_iterate(&destiterator, req, "c");
		if (!ast_strlen_zero(c)) {
			if (sscanf(c, "IN IP4 %256s", host) != 1) {
				ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c);
			} else {
				/* XXX This could block for a long time, and block the main thread! XXX */
				if (audio) {
					if ( !(hp = ast_gethostbyname(host, &audiohp)))
						ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in secondary c= line, '%s'\n", c);
				} else if (!(vhp = ast_gethostbyname(host, &videohp)))
					ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
			}

		}
	}
	if (portno == -1 && vportno == -1 && udptlportno == -1)
		/* No acceptable offer found in SDP  - we have no ports */
		/* Do not change RTP or VRTP if this is a re-invite */
		return -2;

	if (numberofmediastreams > 2)
		/* We have too many fax, audio and/or video media streams, fail this offer */
		return -3;

	/* RTP addresses and ports for audio and video */
	sin.sin_family = AF_INET;
	vsin.sin_family = AF_INET;
	memcpy(&sin.sin_addr, hp->h_addr, sizeof(sin.sin_addr));
	if (vhp)
		memcpy(&vsin.sin_addr, vhp->h_addr, sizeof(vsin.sin_addr));

	/* Setup UDPTL port number */
	if (p->udptl) {
		if (udptlportno > 0) {
			sin.sin_port = htons(udptlportno);
			ast_udptl_set_peer(p->udptl, &sin);
			if (debug)
				ast_log(LOG_DEBUG,"Peer T.38 UDPTL is at port %s:%d\n",ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
		} else {
			ast_udptl_stop(p->udptl);
			if (debug)
				ast_log(LOG_DEBUG, "Peer doesn't provide T.38 UDPTL\n");
		}
	}

		
	if (p->rtp) {
		if (portno > 0) {
			sin.sin_port = htons(portno);
			ast_rtp_set_peer(p->rtp, &sin);
			if (debug)
				ast_verbose("Peer audio RTP is at port %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
		} else {
			if (udptlportno > 0) {
				if (debug)
					ast_verbose("Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid %s\n", p->callid);
			} else {
				ast_rtp_stop(p->rtp);
				if (debug)
					ast_verbose("Peer doesn't provide audio. Callid %s\n", p->callid);
			}
		}
	}
	/* Setup video port number */
	if (vportno != -1)
		vsin.sin_port = htons(vportno);

	/* Next, scan through each "a=rtpmap:" line, noting each
	 * specified RTP payload type (with corresponding MIME subtype):
	 */
	/* XXX This needs to be done per media stream, since it's media stream specific */
	iterator = req->sdp_start;
	while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
		char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
		if (option_debug > 1) {
			int breakout = FALSE;
		
			/* If we're debugging, check for unsupported sdp options */
			if (!strncasecmp(a, "rtcp:", (size_t) 5)) {
				if (debug)
					ast_verbose("Got unsupported a:rtcp in SDP offer \n");
				breakout = TRUE;
			} else if (!strncasecmp(a, "fmtp:", (size_t) 5)) {
				/* Format parameters:  Not supported */
				/* Note: This is used for codec parameters, like bitrate for
					G722 and video formats for H263 and H264 
					See RFC2327 for an example */
				if (debug)
					ast_verbose("Got unsupported a:fmtp in SDP offer \n");
				breakout = TRUE;
			} else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
				/* Video stuff:  Not supported */
				if (debug)
					ast_verbose("Got unsupported a:framerate in SDP offer \n");
				breakout = TRUE;
			} else if (!strncasecmp(a, "maxprate:", (size_t) 9)) {
				/* Video stuff:  Not supported */
				if (debug)
					ast_verbose("Got unsupported a:maxprate in SDP offer \n");
				breakout = TRUE;
			} else if (!strncasecmp(a, "crypto:", (size_t) 7)) {
				/* SRTP stuff, not yet supported */
				if (debug)
					ast_verbose("Got unsupported a:crypto in SDP offer \n");
				breakout = TRUE;
			}