chan_sip.c

	 * specified RTP payload type (with corresponding MIME subtype):
	 */
	iterator = 0;
	while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
		char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
		if (!strcasecmp(a, "sendonly")) {
			sendonly=1;
			continue;
		}
		if (!strcasecmp(a, "sendrecv")) {
		  	sendonly=0;
		}
		if (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) != 2) continue;
		if (debug)
			ast_verbose("Found description format %s\n", mimeSubtype);
		/* Note: should really look at the 'freq' and '#chans' params too */
		ast_rtp_set_rtpmap_type(p->rtp, codec, "audio", mimeSubtype);
		if (p->vrtp)
			ast_rtp_set_rtpmap_type(p->vrtp, codec, "video", mimeSubtype);
	}

	/* Now gather all of the codecs that were asked for: */
	ast_rtp_get_current_formats(p->rtp,
				&peercapability, &peernoncodeccapability);
	if (p->vrtp)
		ast_rtp_get_current_formats(p->vrtp,
				&vpeercapability, &vpeernoncodeccapability);
	p->jointcapability = p->capability & (peercapability | vpeercapability);
	p->peercapability = (peercapability | vpeercapability);
	p->noncodeccapability = noncodeccapability & peernoncodeccapability;
	
	if (ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_AUTO) {
		ast_clear_flag(&p->flags[0], SIP_DTMF);
		if (p->noncodeccapability & AST_RTP_DTMF) {
			/* XXX Would it be reasonable to drop the DSP at this point? XXX */
			ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
		} else {
			ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
		}
	}
	
	if (debug) {
		/* shame on whoever coded this.... */
		const unsigned slen=512;
		char s1[slen], s2[slen], s3[slen], s4[slen];

		ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n",
			ast_getformatname_multiple(s1, slen, p->capability),
			ast_getformatname_multiple(s2, slen, peercapability),
			ast_getformatname_multiple(s3, slen, vpeercapability),
			ast_getformatname_multiple(s4, slen, p->jointcapability));

		ast_verbose("Non-codec capabilities: us - %s, peer - %s, combined - %s\n",
			ast_rtp_lookup_mime_multiple(s1, slen, noncodeccapability, 0),
			ast_rtp_lookup_mime_multiple(s2, slen, peernoncodeccapability, 0),
			ast_rtp_lookup_mime_multiple(s3, slen, p->noncodeccapability, 0));
	}
	if (!p->jointcapability) {
		ast_log(LOG_NOTICE, "No compatible codecs!\n");
		return -1;
	}

	if (!p->owner) 	/* There's no open channel owning us */
		return 0;

	if (!(p->owner->nativeformats & p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
		const unsigned slen=512;
		char s1[slen], s2[slen];
		ast_log(LOG_DEBUG, "Oooh, we need to change our formats since our peer supports only %s and not %s\n", 
				ast_getformatname_multiple(s1, slen, p->jointcapability),
				ast_getformatname_multiple(s2, slen, p->owner->nativeformats));
		p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability);
		ast_set_read_format(p->owner, p->owner->readformat);
		ast_set_write_format(p->owner, p->owner->writeformat);
	}
	if ((bridgepeer=ast_bridged_channel(p->owner))) {
		/* We have a bridge */
		/* Turn on/off music on hold if we are holding/unholding */
		if (sin.sin_addr.s_addr && !sendonly) {
			ast_moh_stop(bridgepeer);
		
			/* Activate a re-invite */
			ast_queue_frame(p->owner, &ast_null_frame);
		} else {
			/* No address for RTP, we're on hold */
			
			ast_moh_start(bridgepeer, NULL);
			if (sendonly)
				ast_rtp_stop(p->rtp);
			/* Activate a re-invite */
			ast_queue_frame(p->owner, &ast_null_frame);
		}
	}

	/* Manager Hold and Unhold events must be generated, if necessary */
	if (sin.sin_addr.s_addr && !sendonly) {
		append_history(p, "Unhold", "%s", req->data);

		if (global_callevents && ast_test_flag(&p->flags[0], SIP_CALL_ONHOLD)) {
			manager_event(EVENT_FLAG_CALL, "Unhold",
				"Channel: %s\r\n"
				"Uniqueid: %s\r\n",
				p->owner->name, 
				p->owner->uniqueid);

		}
		ast_clear_flag(&p->flags[0], SIP_CALL_ONHOLD);
	} else {
		/* No address for RTP, we're on hold */
		append_history(p, "Hold", "%s", req->data);

		if (global_callevents && !ast_test_flag(&p->flags[0], SIP_CALL_ONHOLD)) {
			manager_event(EVENT_FLAG_CALL, "Hold",
				"Channel: %s\r\n"
				"Uniqueid: %s\r\n",
				p->owner->name, 
				p->owner->uniqueid);
		}
		ast_set_flag(&p->flags[0], SIP_CALL_ONHOLD);
	}

	return 0;
}

/*! \brief Add header to SIP message */
static int add_header(struct sip_request *req, const char *var, const char *value)
{
	int maxlen = sizeof(req->data) - 4 - req->len; /* 4 bytes are for two \r\n ? */

	if (req->headers == SIP_MAX_HEADERS) {
		ast_log(LOG_WARNING, "Out of SIP header space\n");
		return -1;
	}

	if (req->lines) {
		ast_log(LOG_WARNING, "Can't add more headers when lines have been added\n");
		return -1;
	}

	if (maxlen <= 0) {
		ast_log(LOG_WARNING, "Out of space, can't add anymore (%s:%s)\n", var, value);
		return -1;
	}

	req->header[req->headers] = req->data + req->len;

	if (compactheaders)
		var = find_alias(var, var);

	snprintf(req->header[req->headers], maxlen, "%s: %s\r\n", var, value);
	req->len += strlen(req->header[req->headers]);
	req->headers++;

	return 0;	
}

/*! \brief Add 'Content-Length' header to SIP message */
static int add_header_contentLength(struct sip_request *req, int len)
{
	char clen[10];

	snprintf(clen, sizeof(clen), "%d", len);
	return add_header(req, "Content-Length", clen);
}

/*! \brief Add content (not header) to SIP message */
static int add_line(struct sip_request *req, const char *line)
{
	if (req->lines == SIP_MAX_LINES)  {
		ast_log(LOG_WARNING, "Out of SIP line space\n");
		return -1;
	}
	if (!req->lines) {
		/* Add extra empty return */
		snprintf(req->data + req->len, sizeof(req->data) - req->len, "\r\n");
		req->len += strlen(req->data + req->len);
	}
	if (req->len >= sizeof(req->data) - 4) {
		ast_log(LOG_WARNING, "Out of space, can't add anymore\n");
		return -1;
	}
	req->line[req->lines] = req->data + req->len;
	snprintf(req->line[req->lines], sizeof(req->data) - req->len, "%s", line);
	req->len += strlen(req->line[req->lines]);
	req->lines++;
	return 0;	
}

/*! \brief Copy one header field from one request to another */
static int copy_header(struct sip_request *req, struct sip_request *orig, char *field)
{
	const char *tmp = get_header(orig, field);

	if (!ast_strlen_zero(tmp)) /* Add what we're responding to */
		return add_header(req, field, tmp);
	ast_log(LOG_NOTICE, "No field '%s' present to copy\n", field);
	return -1;
}

/*! \brief Copy all headers from one request to another */
static int copy_all_header(struct sip_request *req, struct sip_request *orig, char *field)
{
	int start = 0;
	int copied = 0;
	for (;;) {
		const char *tmp = __get_header(orig, field, &start);

		if (ast_strlen_zero(tmp))
			break;
		/* Add what we're responding to */
		add_header(req, field, tmp);
		copied++;
	}
	return copied ? 0 : -1;
}

/*! \brief Copy SIP VIA Headers from the request to the response
\note	If the client indicates that it wishes to know the port we received from,
	it adds ;rport without an argument to the topmost via header. We need to
	add the port number (from our point of view) to that parameter.
	We always add ;received=<ip address> to the topmost via header.
	Received: RFC 3261, rport RFC 3581 */
static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, struct sip_request *orig, char *field)
{
	int copied = 0;
	int start = 0;

	for (;;) {
		char new[256];
		const char *oh = __get_header(orig, field, &start);

		if (ast_strlen_zero(oh))
			break;

		if (!copied) {	/* Only check for empty rport in topmost via header */
			char iabuf[INET_ADDRSTRLEN];
			char *rport;

			/* Find ;rport;  (empty request) */
			rport = strstr(oh, ";rport");
			if (rport && *(rport+6) == '=') 
				rport = NULL;		/* We already have a parameter to rport */

			if (rport && ast_test_flag(&p->flags[0], SIP_NAT) == SIP_NAT_ALWAYS) {
				/* We need to add received port - rport */
				char tmp[256], *end;

				ast_copy_string(tmp, oh, sizeof(tmp));

				rport = strstr(tmp, ";rport");

				if (rport) {
					end = strchr(rport + 1, ';');
					if (end)
						memmove(rport, end, strlen(end) + 1);
					else
						*rport = '\0';
				}

				/* Add rport to first VIA header if requested */
				/* Whoo hoo!  Now we can indicate port address translation too!  Just
				   another RFC (RFC3581). I'll leave the original comments in for
				   posterity.  */
				snprintf(new, sizeof(new), "%s;received=%s;rport=%d",
					tmp, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr),
					ntohs(p->recv.sin_port));
			} else {
				/* We should *always* add a received to the topmost via */
				snprintf(new, sizeof(new), "%s;received=%s",
					oh, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr));
			}
			oh = new;	/* the header to copy */
		}  /* else add the following via headers untouched */
		add_header(req, field, oh);
		copied++;
	}
	if (!copied) {
		ast_log(LOG_NOTICE, "No header field '%s' present to copy\n", field);
		return -1;
	}
	return 0;
}

/*! \brief Add route header into request per learned route */
static void add_route(struct sip_request *req, struct sip_route *route)
{
	char r[256], *p;
	int n, rem = sizeof(r);

	if (!route)
		return;

	p = r;
	for (;route ; route = route->next) {
		n = strlen(route->hop);
		if (rem < n+3) /* we need room for ",<route>" */
			break;
		if (p != r) {	/* add a separator after fist route */
			*p++ = ',';
			--rem;
		}
		*p++ = '<';
		ast_copy_string(p, route->hop, rem); /* cannot fail */
		p += n;
		*p++ = '>';
		rem -= (n+2);
	}
	*p = '\0';
	add_header(req, "Route", r);
}

/*! \brief Set destination from SIP URI */
static void set_destination(struct sip_pvt *p, char *uri)
{
	char *h, *maddr, hostname[256];
	char iabuf[INET_ADDRSTRLEN];
	int port, hn;
	struct hostent *hp;
	struct ast_hostent ahp;
	int debug=sip_debug_test_pvt(p);

	/* Parse uri to h (host) and port - uri is already just the part inside the <> */
	/* general form we are expecting is sip[s]:username[:password]@host[:port][;...] */

	if (debug)
		ast_verbose("set_destination: Parsing <%s> for address/port to send to\n", uri);

	/* Find and parse hostname */
	h = strchr(uri, '@');
	if (h)
		++h;
	else {
		h = uri;
		if (strncmp(h, "sip:", 4) == 0)
			h += 4;
		else if (strncmp(h, "sips:", 5) == 0)
			h += 5;
	}
	hn = strcspn(h, ":;>") + 1;
	if (hn > sizeof(hostname)) 
		hn = sizeof(hostname);
	ast_copy_string(hostname, h, hn);
	/* XXX bug here if string has been trimmed to sizeof(hostname) */
	h += hn - 1;

	/* Is "port" present? if not default to DEFAULT_SIP_PORT */
	if (*h == ':') {
		/* Parse port */
		++h;
		port = strtol(h, &h, 10);
	}
	else
		port = DEFAULT_SIP_PORT;

	/* Got the hostname:port - but maybe there's a "maddr=" to override address? */
	maddr = strstr(h, "maddr=");
	if (maddr) {
		maddr += 6;
		hn = strspn(maddr, "0123456789.") + 1;
		if (hn > sizeof(hostname))
			hn = sizeof(hostname);
		ast_copy_string(hostname, maddr, hn);
	}
	
	hp = ast_gethostbyname(hostname, &ahp);
	if (hp == NULL)  {
		ast_log(LOG_WARNING, "Can't find address for host '%s'\n", hostname);
		return;
	}
	p->sa.sin_family = AF_INET;
	memcpy(&p->sa.sin_addr, hp->h_addr, sizeof(p->sa.sin_addr));
	p->sa.sin_port = htons(port);
	if (debug)
		ast_verbose("set_destination: set destination to %s, port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), port);
}

/*! \brief Initialize SIP response, based on SIP request */
static int init_resp(struct sip_request *resp, const char *msg)
{
	/* Initialize a response */
	memset(resp, 0, sizeof(*resp));
	resp->method = SIP_RESPONSE;
	resp->header[0] = resp->data;
	snprintf(resp->header[0], sizeof(resp->data), "SIP/2.0 %s\r\n", msg);
	resp->len = strlen(resp->header[0]);
	resp->headers++;
	return 0;
}

/*! \brief Initialize SIP request */
static int init_req(struct sip_request *req, int sipmethod, const char *recip)
{
	/* Initialize a request */
	memset(req, 0, sizeof(*req));
        req->method = sipmethod;
	req->header[0] = req->data;
	snprintf(req->header[0], sizeof(req->data), "%s %s SIP/2.0\r\n", sip_methods[sipmethod].text, recip);
	req->len = strlen(req->header[0]);
	req->headers++;
	return 0;
}


/*! \brief Prepare SIP response packet */
static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, struct sip_request *req)
{
	char newto[256];
	const char *ot;

	init_resp(resp, msg);
	copy_via_headers(p, resp, req, "Via");
	if (msg[0] == '2')
		copy_all_header(resp, req, "Record-Route");
	copy_header(resp, req, "From");
	ot = get_header(req, "To");
	if (!strcasestr(ot, "tag=") && strncmp(msg, "100", 3)) {
		/* Add the proper tag if we don't have it already.  If they have specified
		   their tag, use it.  Otherwise, use our own tag */
		if (!ast_strlen_zero(p->theirtag) && ast_test_flag(&p->flags[0], SIP_OUTGOING))
			snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
		else if (p->tag && !ast_test_flag(&p->flags[0], SIP_OUTGOING))
			snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
		else
			ast_copy_string(newto, ot, sizeof(newto));
		ot = newto;
	}
	add_header(resp, "To", ot);
	copy_header(resp, req, "Call-ID");
	copy_header(resp, req, "CSeq");
	add_header(resp, "User-Agent", global_useragent);
	add_header(resp, "Allow", ALLOWED_METHODS);
	add_header(resp, "Supported", SUPPORTED_EXTENSIONS);
	if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER)) {
		/* For registration responses, we also need expiry and
		   contact info */
		char tmp[256];

		snprintf(tmp, sizeof(tmp), "%d", p->expiry);
		add_header(resp, "Expires", tmp);
		if (p->expiry) {	/* Only add contact if we have an expiry time */
			char contact[256];
			snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry);
			add_header(resp, "Contact", contact);	/* Not when we unregister */
		}
	} else if (p->our_contact[0]) {
		add_header(resp, "Contact", p->our_contact);
	}
	return 0;
}

/*! \brief Initialize a SIP request response packet */
static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch)
{
	struct sip_request *orig = &p->initreq;
	char stripped[80];
	char tmp[80];
	char newto[256];
	const char *c;
	char *n;
	const char *ot, *of;
	int is_strict = FALSE;		/*!< Strict routing flag */

	memset(req, 0, sizeof(struct sip_request));
	
	snprintf(p->lastmsg, sizeof(p->lastmsg), "Tx: %s", sip_methods[sipmethod].text);
	
	if (!seqno) {
		p->ocseq++;
		seqno = p->ocseq;
	}
	
	if (newbranch) {
		p->branch ^= ast_random();
		build_via(p);
	}

	/* Check for strict or loose router */
	if (p->route && !ast_strlen_zero(p->route->hop) && strstr(p->route->hop,";lr") == NULL) {
		is_strict = TRUE;
		if (sipdebug)
			ast_log(LOG_DEBUG, "Strict routing enforced for session %s\n", p->callid);
	}

	if (sipmethod == SIP_CANCEL) {
		c = p->initreq.rlPart2;	/* Use original URI */
	} else if (sipmethod == SIP_ACK) {
		/* Use URI from Contact: in 200 OK (if INVITE) 
		(we only have the contacturi on INVITEs) */
		if (!ast_strlen_zero(p->okcontacturi))
			c = is_strict ? p->route->hop : p->okcontacturi;
 		else
 			c = p->initreq.rlPart2;
	} else if (!ast_strlen_zero(p->okcontacturi)) {
		c = is_strict ? p->route->hop : p->okcontacturi; /* Use for BYE or REINVITE */
	} else if (!ast_strlen_zero(p->uri)) {
		c = p->uri;
	} else {
		/* We have no URI, use To: or From:  header as URI (depending on direction) */
		ast_copy_string(stripped, get_header(orig, (ast_test_flag(&p->flags[0], SIP_OUTGOING)) ? "To" : "From"),
				sizeof(stripped));
		c = get_in_brackets(stripped);
		n = strchr(c, ';');
		if (n)
			*n = '\0';
	}	
	init_req(req, sipmethod, c);

	snprintf(tmp, sizeof(tmp), "%d %s", seqno, sip_methods[sipmethod].text);

	add_header(req, "Via", p->via);
	if (p->route) {
		set_destination(p, p->route->hop);
		add_route(req, is_strict ? p->route->next : p->route);
	}

	ot = get_header(orig, "To");
	of = get_header(orig, "From");

	/* Add tag *unless* this is a CANCEL, in which case we need to send it exactly
	   as our original request, including tag (or presumably lack thereof) */
	if (!strcasestr(ot, "tag=") && sipmethod != SIP_CANCEL) {
		/* Add the proper tag if we don't have it already.  If they have specified
		   their tag, use it.  Otherwise, use our own tag */
		if (ast_test_flag(&p->flags[0], SIP_OUTGOING) && !ast_strlen_zero(p->theirtag))
			snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->theirtag);
		else if (!ast_test_flag(&p->flags[0], SIP_OUTGOING))
			snprintf(newto, sizeof(newto), "%s;tag=%s", ot, p->tag);
		else
			snprintf(newto, sizeof(newto), "%s", ot);
		ot = newto;
	}

	if (ast_test_flag(&p->flags[0], SIP_OUTGOING)) {
		add_header(req, "From", of);
		add_header(req, "To", ot);
	} else {
		add_header(req, "From", ot);
		add_header(req, "To", of);
	}
	add_header(req, "Contact", p->our_contact);
	copy_header(req, orig, "Call-ID");
	add_header(req, "CSeq", tmp);

	add_header(req, "User-Agent", global_useragent);
	add_header(req, "Max-Forwards", DEFAULT_MAX_FORWARDS);

	if (!ast_strlen_zero(p->rpid))
		add_header(req, "Remote-Party-ID", p->rpid);

	return 0;
}

/*! \brief Base transmit response function */
static int __transmit_response(struct sip_pvt *p, const char *msg, struct sip_request *req, enum xmittype reliable)
{
	struct sip_request resp;
	int seqno = 0;

	if (reliable && (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1)) {
		ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
		return -1;
	}
	respprep(&resp, p, msg, req);
	add_header_contentLength(&resp, 0);
	/* If we are cancelling an incoming invite for some reason, add information
		about the reason why we are doing this in clear text */
	if (p->method == SIP_INVITE && msg[0] != '1' && p->owner && p->owner->hangupcause) {
		char buf[10];

		add_header(&resp, "X-Asterisk-HangupCause", ast_cause2str(p->owner->hangupcause));
		snprintf(buf, sizeof(buf), "%d", p->owner->hangupcause);
		add_header(&resp, "X-Asterisk-HangupCauseCode", buf);
	}
	return send_response(p, &resp, reliable, seqno);
}

/*! \brief Transmit response, no retransmits */
static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req) 
{
	return __transmit_response(p, msg, req, XMIT_UNRELIABLE);
}

/*! \brief Transmit response, no retransmits */
static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *unsupported) 
{
	struct sip_request resp;
	respprep(&resp, p, msg, req);
	append_date(&resp);
	add_header(&resp, "Unsupported", unsupported);
	return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

/*! \brief Transmit response, Make sure you get an ACK
	This is only used for responses to INVITEs, where we need to make sure we get an ACK
*/
static int transmit_response_reliable(struct sip_pvt *p, const char *msg, struct sip_request *req)
{
	return __transmit_response(p, msg, req, XMIT_CRITICAL);
}

/*! \brief Append date to SIP message */
static void append_date(struct sip_request *req)
{
	char tmpdat[256];
	struct tm tm;
	time_t t = time(NULL);

	gmtime_r(&t, &tm);
	strftime(tmpdat, sizeof(tmpdat), "%a, %d %b %Y %T GMT", &tm);
	add_header(req, "Date", tmpdat);
}

/*! \brief Append date and content length before transmitting response */
static int transmit_response_with_date(struct sip_pvt *p, char *msg, struct sip_request *req)
{
	struct sip_request resp;
	respprep(&resp, p, msg, req);
	append_date(&resp);
	add_header_contentLength(&resp, 0);
	return send_response(p, &resp, XMIT_UNRELIABLE, 0);
}

/*! \brief Append Accept header, content length before transmitting response */
static int transmit_response_with_allow(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable)
{
	struct sip_request resp;
	respprep(&resp, p, msg, req);
	add_header(&resp, "Accept", "application/sdp");
	add_header_contentLength(&resp, 0);
	return send_response(p, &resp, reliable, 0);
}

/*! \brief Respond with authorization request */
static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, struct sip_request *req, const char *randdata, enum xmittype reliable, const char *header, int stale)
{
	struct sip_request resp;
	char tmp[256];
	int seqno = 0;

	if (reliable && (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1)) {
		ast_log(LOG_WARNING, "Unable to determine sequence number from '%s'\n", get_header(req, "CSeq"));
		return -1;
	}
	/* Stale means that they sent us correct authentication, but 
	   based it on an old challenge (nonce) */
	snprintf(tmp, sizeof(tmp), "Digest realm=\"%s\", nonce=\"%s\"%s", global_realm, randdata, stale ? ", stale=true" : "");
	respprep(&resp, p, msg, req);
	add_header(&resp, header, tmp);
	add_header_contentLength(&resp, 0);
	return send_response(p, &resp, reliable, seqno);
}

/*! \brief Add text body to SIP message */
static int add_text(struct sip_request *req, const char *text)
{
	/* XXX Convert \n's to \r\n's XXX */
	add_header(req, "Content-Type", "text/plain");
	add_header_contentLength(req, strlen(text));
	add_line(req, text);
	return 0;
}

/*! \brief Add DTMF INFO tone to sip message */
/* Always adds default duration 250 ms, regardless of what came in over the line */
static int add_digit(struct sip_request *req, char digit)
{
	char tmp[256];

	snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=250\r\n", digit);
	add_header(req, "Content-Type", "application/dtmf-relay");
	add_header_contentLength(req, strlen(tmp));
	add_line(req, tmp);
	return 0;
}

/*! \brief add XML encoded media control with update */
/*! \note XML: The only way to turn 0 bits of information into a few hundred. */
static int add_vidupdate(struct sip_request *req)
{
	const char *xml_is_a_huge_waste_of_space =
		"<?xml version=\"1.0\" encoding=\"utf-8\" ?>\r\n"
		" <media_control>\r\n"
		"  <vc_primitive>\r\n"
		"   <to_encoder>\r\n"
		"    <picture_fast_update>\r\n"
		"    </picture_fast_update>\r\n"
		"   </to_encoder>\r\n"
		"  </vc_primitive>\r\n"
		" </media_control>\r\n";
	add_header(req, "Content-Type", "application/media_control+xml");
	add_header_contentLength(req, strlen(xml_is_a_huge_waste_of_space));
	add_line(req, xml_is_a_huge_waste_of_space);
	return 0;
}

static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
			     char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
			     int debug)
{
	int rtp_code;

	if (debug)
		ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
	if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1)
		return;

	ast_build_string(m_buf, m_size, " %d", rtp_code);
	ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
			 ast_rtp_lookup_mime_subtype(1, codec),
			 sample_rate);
	if (codec == AST_FORMAT_G729A)
		/* Indicate that we don't support VAD (G.729 annex B) */
		ast_build_string(a_buf, a_size, "a=fmtp:%d annexb=no\r\n", rtp_code);
}

static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
				char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
				int debug)
{
	int rtp_code;

	if (debug)
		ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format));
	if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1)
		return;

	ast_build_string(m_buf, m_size, " %d", rtp_code);
	ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
			 ast_rtp_lookup_mime_subtype(0, format),
			 sample_rate);
	if (format == AST_RTP_DTMF)
		/* Indicate we support DTMF and FLASH... */
		ast_build_string(a_buf, a_size, "a=fmtp:%d 0-16\r\n", rtp_code);
}

/*! \brief Add Session Description Protocol message */
static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
{
	int len = 0;
	int pref_codec;
	int alreadysent = 0;
	struct sockaddr_in sin;
	struct sockaddr_in vsin;
	char v[256];
	char s[256];
	char o[256];
	char c[256];
	char t[256];
	char b[256];
	char *hold;
	char m_audio[256];
	char m_video[256];
	char a_audio[1024];
	char a_video[1024];
	char *m_audio_next = m_audio;
	char *m_video_next = m_video;
	size_t m_audio_left = sizeof(m_audio);
	size_t m_video_left = sizeof(m_video);
	char *a_audio_next = a_audio;
	char *a_video_next = a_video;
	size_t a_audio_left = sizeof(a_audio);
	size_t a_video_left = sizeof(a_video);
	char iabuf[INET_ADDRSTRLEN];
	int x;
	int capability;
	struct sockaddr_in dest;
	struct sockaddr_in vdest = { 0, };
	int debug;
	
	debug = sip_debug_test_pvt(p);

	len = 0;
	if (!p->rtp) {
		ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
		return -1;
	}
	capability = p->jointcapability;
		
	if (!p->sessionid) {
		p->sessionid = getpid();
		p->sessionversion = p->sessionid;
	} else
		p->sessionversion++;
	ast_rtp_get_us(p->rtp, &sin);
	if (p->vrtp)
		ast_rtp_get_us(p->vrtp, &vsin);

	if (p->redirip.sin_addr.s_addr) {
		dest.sin_port = p->redirip.sin_port;
		dest.sin_addr = p->redirip.sin_addr;
		if (p->redircodecs)
			capability = p->redircodecs;
	} else {
		dest.sin_addr = p->ourip;
		dest.sin_port = sin.sin_port;
	}

	/* Determine video destination */
	if (p->vrtp) {
		if (p->vredirip.sin_addr.s_addr) {
			vdest.sin_port = p->vredirip.sin_port;
			vdest.sin_addr = p->vredirip.sin_addr;
		} else {
			vdest.sin_addr = p->ourip;
			vdest.sin_port = vsin.sin_port;
		}
	}
	if (debug) {
		ast_verbose("We're at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(sin.sin_port));	
		if (p->vrtp)
			ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(vsin.sin_port));	
	}

	/* We break with the "recommendation" and send our IP, in order that our
	   peer doesn't have to ast_gethostbyname() us */

	snprintf(v, sizeof(v), "v=0\r\n");
	snprintf(o, sizeof(o), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr));
	snprintf(s, sizeof(s), "s=session\r\n");
	snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr));
	if ((p->vrtp) &&
	    (!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
	    (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
		snprintf(b, sizeof(b), "b=CT:%d\r\n", p->maxcallbitrate);	
	snprintf(t, sizeof(t), "t=0 0\r\n");

	ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
	ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port));

	if (ast_test_flag(&p->flags[0], SIP_CALL_ONHOLD))
		hold = "a=recvonly";
	else
		hold = "a=sendrecv";

	/* Prefer the codec we were requested to use, first, no matter what */
	if (capability & p->prefcodec) {
		if (p->prefcodec <= AST_FORMAT_MAX_AUDIO)
			add_codec_to_sdp(p, p->prefcodec, 8000,
					 &m_audio_next, &m_audio_left,
					 &a_audio_next, &a_audio_left,
					 debug);
		else
			add_codec_to_sdp(p, p->prefcodec, 90000,
					 &m_video_next, &m_video_left,
					 &a_video_next, &a_video_left,
					 debug);
		alreadysent |= p->prefcodec;
	}

	/* Start by sending our preferred codecs */
	for (x = 0; x < 32; x++) {
		if (!(pref_codec = ast_codec_pref_index(&p->prefs, x)))
			break; 

		if (!(capability & pref_codec))
			continue;

		if (alreadysent & pref_codec)
			continue;

		if (pref_codec <= AST_FORMAT_MAX_AUDIO)
			add_codec_to_sdp(p, pref_codec, 8000,
					 &m_audio_next, &m_audio_left,
					 &a_audio_next, &a_audio_left,
					 debug);
		else
			add_codec_to_sdp(p, pref_codec, 90000,
					 &m_video_next, &m_video_left,
					 &a_video_next, &a_video_left,
					 debug);
		alreadysent |= pref_codec;
	}

	/* Now send any other common codecs, and non-codec formats: */
	for (x = 1;
	     x <= ((ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO);
	     x <<= 1) {
		if (!(capability & x))
			continue;

		if (alreadysent & x)
			continue;

		if (x <= AST_FORMAT_MAX_AUDIO)
			add_codec_to_sdp(p, x, 8000,
					 &m_audio_next, &m_audio_left,
					 &a_audio_next, &a_audio_left,
					 debug);
		else
			add_codec_to_sdp(p, x, 90000,
					 &m_video_next, &m_video_left,
					 &a_video_next, &a_video_left,
					 debug);
	}

	for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
		if (!(p->noncodeccapability & x))
			continue;

		add_noncodec_to_sdp(p, x, 8000,
				    &m_audio_next, &m_audio_left,
				    &a_audio_next, &a_audio_left,
				    debug);
	}

	if(!ast_internal_timing_enabled(p->owner))
		ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n");

	if ((m_audio_left < 2) || (m_video_left < 2) || (a_audio_left == 0) || (a_video_left == 0))
		ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");

	ast_build_string(&m_audio_next, &m_audio_left, "\r\n");
	ast_build_string(&m_video_next, &m_video_left, "\r\n");

	len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_audio) + strlen(a_audio) + strlen(hold);
	if ((p->vrtp) &&
	    (!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
	    (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
		len += strlen(m_video) + strlen(a_video) + strlen(b) + strlen(hold);

	add_header(resp, "Content-Type", "application/sdp");
	add_header_contentLength(resp, len);
	add_line(resp, v);
	add_line(resp, o);
	add_line(resp, s);
	add_line(resp, c);
	if ((p->vrtp) &&
	    (!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
	    (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
		add_line(resp, b);
	add_line(resp, t);
	add_line(resp, m_audio);
	add_line(resp, a_audio);
	add_line(resp, hold);
	if ((p->vrtp) &&
	    (!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
	    (capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */
		add_line(resp, m_video);
		add_line(resp, a_video);
		add_line(resp, hold);
	}

	/* Update lastrtprx when we send our SDP */
	time(&p->lastrtprx);
	time(&p->lastrtptx);

	return 0;
}

/*! \brief copy SIP request (mostly used to save request for responses) */
static void copy_request(struct sip_request *dst, struct sip_request *src)
{
	long offset;
	int x;
	offset = ((void *)dst) - ((void *)src);
	/* First copy stuff */
	memcpy(dst, src, sizeof(*dst));
	/* Now fix pointer arithmetic */
	for (x=0; x < src->headers; x++)
		dst->header[x] += offset;
	for (x=0; x < src->lines; x++)
		dst->line[x] += offset;
}

/*! \brief Used for 200 OK and 183 early media */
static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, enum xmittype reliable)
{
	struct sip_request resp;
	int seqno;
	if (sscanf(get_header(req, "CSeq"), "%d ", &seqno) != 1) {
		ast_log(LOG_WARNING, "Unable to get seqno from '%s'\n", get_header(req, "CSeq"));
		return -1;
	}
	respprep(&resp, p, msg, req);
	if (p->rtp) {
		try_suggested_sip_codec(p);	
		add_sdp(&resp, p);
	} else {
		ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
	}
	return send_response(p, &resp, reliable, seqno);
}

/*! \brief Parse first line of incoming SIP request */
static int determine_firstline_parts(struct sip_request *req) 
{
	char *e = ast_skip_blanks(req->header[0]);	/* there shouldn't be any */

	if (!*e)
		return -1;
	req->rlPart1 = e;	/* method or protocol */
	e = ast_skip_nonblanks(e);
	if (*e)
		*e++ = '\0';
	/* Get URI or status code */
	e = ast_skip_blanks(e);
	if ( !*e )
		return -1;
	ast_trim_blanks(e);