/*
* PROXY_REWRITE_INVITATION_BODY
*
* rewrites the outgoing INVITATION request or response packet
*
* RETURNS
* STS_SUCCESS on success
* STS_FAILURE on error
*/
int proxy_rewrite_invitation_body(osip_message_t *mymsg, int direction){
osip_body_t *body;
sdp_message_t *sdp;
struct in_addr map_addr, addr_sess, addr_media, outside_addr, inside_addr;
int sts;
char *bodybuff;
int bodybuflen;
char clen[8]; /* content length: probably never more than 7 digits !*/
int map_port, msg_port;
int media_stream_no;
sdp_connection_t *sdp_conn;
sdp_media_t *sdp_med;
int rtp_direction=0;
int have_c_media=0;
if (configuration.rtp_proxy_enable == 0) return STS_SUCCESS;
/*
* get SDP structure
*/
sts = osip_message_get_body(mymsg, 0, &body);
if (sts != 0) {
#if 0
if ((MSG_IS_RESPONSE_FOR(mymsg,"INVITE")) &&
(MSG_IS_STATUS_1XX(mymsg))) {
/* 1xx responses *MAY* contain SDP data */
DEBUGC(DBCLASS_PROXY, "rewrite_invitation_body: "
"no body found in message");
return STS_SUCCESS;
} else {
/* INVITE request and 200 response *MUST* contain SDP data */
ERROR("rewrite_invitation_body: no body found in message");
return STS_FAILURE;
}
#else
DEBUGC(DBCLASS_PROXY, "rewrite_invitation_body: "
"no body found in message");
return STS_SUCCESS;
#endif
}
sts = sip_body_to_str(body, &bodybuff, &bodybuflen);
if (sts != 0) {
ERROR("rewrite_invitation_body: unable to sip_body_to_str");
}
sts = sdp_message_init(&sdp);
sts = sdp_message_parse (sdp, bodybuff);
if (sts != 0) {
ERROR("rewrite_invitation_body: unable to sdp_message_parse body");
DUMP_BUFFER(-1, bodybuff, bodybuflen);
osip_free(bodybuff);
sdp_message_free(sdp);
return STS_FAILURE;
}
osip_free(bodybuff);
if (configuration.debuglevel)
{ /* just dump the buffer */
char *tmp, *tmp2;
int tmplen;
sts = osip_message_get_body(mymsg, 0, &body);
sts = sip_body_to_str(body, &tmp, &tmplen);
osip_content_length_to_str(mymsg->content_length, &tmp2);
DEBUG("Body before rewrite (clen=%s, strlen=%i):\n%s\n----",
tmp2, tmplen, tmp);
osip_free(tmp);
osip_free(tmp2);
}
/*
* RTP proxy: get ready and start forwarding
* start forwarding for each media stream ('m=' item in SIP message)
*/
/* get outbound address */
if (get_interface_ip(IF_OUTBOUND, &outside_addr) != STS_SUCCESS) {
sdp_message_free(sdp);
return STS_FAILURE;
}
/* get inbound address */
if (get_interface_ip(IF_INBOUND, &inside_addr) != STS_SUCCESS) {
sdp_message_free(sdp);
return STS_FAILURE;
}
/* figure out what address to use for RTP masquerading */
if (MSG_IS_REQUEST(mymsg)) {
if (direction == DIR_INCOMING) {
memcpy(&map_addr, &inside_addr, sizeof (map_addr));
rtp_direction = DIR_OUTGOING;
} else {
memcpy(&map_addr, &outside_addr, sizeof (map_addr));
rtp_direction = DIR_INCOMING;
}
} else /* MSG_IS_REPONSE(mymsg) */ {
if (direction == DIR_INCOMING) {
memcpy(&map_addr, &inside_addr, sizeof (map_addr));
rtp_direction = DIR_OUTGOING;
} else {
memcpy(&map_addr, &outside_addr, sizeof (map_addr));
rtp_direction = DIR_INCOMING;
}
}
DEBUGC(DBCLASS_PROXY, "proxy_rewrite_invitation_body: SIP[%s %s] RTP[%s %s]",
MSG_IS_REQUEST(mymsg)? "RQ" : "RS",
(direction==DIR_INCOMING)? "IN" : "OUT",
(rtp_direction==DIR_INCOMING)? "IN" : "OUT",
utils_inet_ntoa(map_addr));
/*
* first, check presence of a 'c=' item on session level
*/
if (sdp->c_connection==NULL || sdp->c_connection->c_addr==NULL) {
/*
* No 'c=' on session level, search on media level now
*
* According to RFC2327, ALL media description must
* include a 'c=' item now:
*/
media_stream_no=0;
while (!sdp_message_endof_media(sdp, media_stream_no)) {
/* check if n'th media stream is present */
if (sdp_message_c_addr_get(sdp, media_stream_no, 0) == NULL) {
ERROR("SDP: have no 'c=' on session level and neither "
"on media level (media=%i)",media_stream_no);
sdp_message_free(sdp);
return STS_FAILURE;
}
media_stream_no++;
} /* while */
}
/* Required 'c=' items ARE present */
/*
* rewrite 'c=' item on session level if present and not yet done.
* remember the original address in addr_sess
*/
memset(&addr_sess, 0, sizeof(addr_sess));
if (sdp->c_connection && sdp->c_connection->c_addr) {
sts = get_ip_by_host(sdp->c_connection->c_addr, &addr_sess);
if (sts == STS_FAILURE) {
ERROR("SDP: cannot resolve session 'c=' host [%s]",
sdp->c_connection->c_addr);
sdp_message_free(sdp);
return STS_FAILURE;
}
/*
* Rewrite
* an IP address of 0.0.0.0 means *MUTE*, don't rewrite such
*/
if (strcmp(sdp->c_connection->c_addr, "0.0.0.0") != 0) {
osip_free(sdp->c_connection->c_addr);
sdp->c_connection->c_addr=osip_malloc(HOSTNAME_SIZE);
sprintf(sdp->c_connection->c_addr, "%s", utils_inet_ntoa(map_addr));
} else {
/* 0.0.0.0 - don't rewrite */
DEBUGC(DBCLASS_PROXY, "proxy_rewrite_invitation_body: "
"got a MUTE c= record (on session level - legal?)");
}
}
/*
* rewrite 'o=' item (originator) on session level if present.
*/
if (sdp->o_addrtype && sdp->o_addr) {
if (strcmp(sdp->o_addrtype, "IP4") != 0) {
ERROR("got IP6 in SDP originator - not yet suported by siproxd");
sdp_message_free(sdp);
return STS_FAILURE;
}
osip_free(sdp->o_addr);
sdp->o_addr=osip_malloc(HOSTNAME_SIZE);
sprintf(sdp->o_addr, "%s", utils_inet_ntoa(map_addr));
}
/*
* loop through all media descritions,
* start RTP proxy and rewrite them
*/
for (media_stream_no=0;;media_stream_no++) {
/* check if n'th media stream is present */
if (sdp_message_m_port_get(sdp, media_stream_no) == NULL) break;
/*
* check if a 'c=' item is present in this media description,
* if so -> rewrite it
*/
memset(&addr_media, 0, sizeof(addr_media));
have_c_media=0;
sdp_conn=sdp_message_connection_get(sdp, media_stream_no, 0);
if (sdp_conn && sdp_conn->c_addr) {
if (strcmp(sdp_conn->c_addr, "0.0.0.0") != 0) {
sts = get_ip_by_host(sdp_conn->c_addr, &addr_media);
have_c_media=1;
/* have a valid address */
osip_free(sdp_conn->c_addr);
sdp_conn->c_addr=osip_malloc(HOSTNAME_SIZE);
sprintf(sdp_conn->c_addr, "%s", utils_inet_ntoa(map_addr));
} else {
/* 0.0.0.0 - don't rewrite */
DEBUGC(DBCLASS_PROXY, "proxy_rewrite_invitation_body: got a "
"MUTE c= record (media level)");
}
}
/* start an RTP proxying stream */
if (sdp_message_m_port_get(sdp, media_stream_no)) {
msg_port=atoi(sdp_message_m_port_get(sdp, media_stream_no));
if (msg_port > 0) {
osip_uri_t *cont_url = NULL;
char *client_id=NULL;
/* try to get some additional UA specific unique ID.
* Try:
* 1) User part of Contact header
* 2) Host part of Contact header (will be different
* between internal UA and external UA)
*/
if (!osip_list_eol(mymsg->contacts, 0))
cont_url = ((osip_contact_t*)(mymsg->contacts->node->element))->url;
if (cont_url) {
client_id=cont_url->username;
if (client_id == NULL) client_id=cont_url->host;
}
/*
* do we have a 'c=' item on media level?
* if not, use the same as on session level
*/
if (have_c_media == 0) {
memcpy(&addr_media, &addr_sess, sizeof(addr_sess));
}
/*
* Am I running in front of the routing device? Then I cannot
* use the external IP to bind a listen socket to, so force
* the use of my inbound IP for listening.
*/
if ((rtp_direction == DIR_INCOMING) &&
(configuration.outbound_host) &&
(strcmp(configuration.outbound_host, "")!=0)) {
DEBUGC(DBCLASS_PROXY, "proxy_rewrite_invitation_body: "
"in-front-of-NAT-Router");
memcpy(&map_addr, &inside_addr, sizeof (map_addr));
}
sts = rtp_start_fwd(osip_message_get_call_id(mymsg),
client_id,
rtp_direction,
media_stream_no,
map_addr, &map_port,
addr_media, msg_port);
if (sts == STS_SUCCESS) {
/* and rewrite the port */
sdp_med=osip_list_get(sdp->m_medias, media_stream_no);
if (sdp_med && sdp_med->m_port) {
osip_free(sdp_med->m_port);
sdp_med->m_port=osip_malloc(8); /* 5 digits, \0 + align */
sprintf(sdp_med->m_port, "%i", map_port);
DEBUGC(DBCLASS_PROXY, "proxy_rewrite_invitation_body: "
"m= rewrote port to [%i]",map_port);
} else {
ERROR("rewriting port in m= failed sdp_med=%p, "
"m_number_of_port=%p", sdp_med, sdp_med->m_port);
}
} /* sts == success */
} /* if msg_port > 0 */
} else {
/* no port defined - skip entry */
WARN("no port defined in m=(media) stream_no=%i", media_stream_no);
continue;
}
} /* for media_stream_no */
/* remove old body */
sts = osip_list_remove(mymsg->bodies, 0);
osip_body_free(body);
/* dump new body */
sdp_message_to_str(sdp, &bodybuff);
bodybuflen=strlen(bodybuff);
/* free sdp structure */
sdp_message_free(sdp);
/* include new body */
sip_message_set_body(mymsg, bodybuff, bodybuflen);
if (sts != 0) {
ERROR("rewrite_invitation_body: unable to sip_message_set_body body");
}
/* free content length resource and include new one*/
osip_content_length_free(mymsg->content_length);
mymsg->content_length=NULL;
sprintf(clen,"%i",bodybuflen);
sts = osip_message_set_content_length(mymsg, clen);
/* free old body */
osip_free(bodybuff);
if (configuration.debuglevel)
{ /* just dump the buffer */
char *tmp, *tmp2;
int tmplen;
sts = osip_message_get_body(mymsg, 0, &body);
sts = sip_body_to_str(body, &tmp, &tmplen);
osip_content_length_to_str(mymsg->content_length, &tmp2);
DEBUG("Body after rewrite (clen=%s, strlen=%i):\n%s\n----",
tmp2, tmplen, tmp);
osip_free(tmp);
osip_free(tmp2);
}
return STS_SUCCESS;
}
/*
* PROXY_RESPONSE
*
* RETURNS
* STS_SUCCESS on success
* STS_FAILURE on error
* RFC3261
* Section 16.7: Proxy Behavior - Response Processing
* 1. Find the appropriate response context
* 2. Update timer C for provisional responses
* 3. Remove the topmost Via
* 4. Add the response to the response context
* 5. Check to see if this response should be forwarded immediately
* 6. When necessary, choose the best final response from the
* response context
* 7. Aggregate authorization header field values if necessary
* 8. Optionally rewrite Record-Route header field values
* 9. Forward the response
* 10. Generate any necessary CANCEL requests
*
*/
int proxy_response (sip_ticket_t *ticket) {
int i;
int sts;
int type;
struct in_addr sendto_addr;
osip_via_t *via;
int port;
char *buffer;
int buflen;
osip_message_t *response;
struct sockaddr_in *from;
DEBUGC(DBCLASS_PROXY,"proxy_response");
if (ticket==NULL) {
ERROR("proxy_response: called with NULL ticket");
return STS_FAILURE;
}
response=ticket->sipmsg;
from=&ticket->from;
/*
* RFC 3261, Section 16.7 step 3
* Proxy Behavior - Response Processing - Remove my Via header field value
*/
/* remove my Via header line */
sts = sip_del_myvia(ticket);
if (sts == STS_FAILURE) {
DEBUGC(DBCLASS_PROXY,"not addressed to my VIA, ignoring response");
return STS_FAILURE;
}
/*
* figure out if this is an request coming from the outside
* world to one of our registered clients
*/
/* Ahhrghh...... a response seems to have NO contact information...
* so let's take FROM instead...
* the TO and FROM headers are EQUAL to the request - that means
* they are swapped in their meaning for a response...
*/
#if _OLD_DIRECTION_EVALUATION
type = 0;
for (i=0; i<URLMAP_SIZE; i++) {
if (urlmap[i].active == 0) continue;
/* incoming response ('from' == 'masq') || ('from' == 'reg') */
if ((compare_url(response->from->url, urlmap[i].reg_url)==STS_SUCCESS) ||
(compare_url(response->from->url, urlmap[i].masq_url)==STS_SUCCESS)) {
type=RESTYP_INCOMING;
DEBUGC(DBCLASS_PROXY,"incoming response for %s@%s from outbound",
response->from->url->username? response->from->url->username:"******",
response->from->url->host? response->from->url->host : "*NULL*");
break;
}
/* outgoing response ('to' == 'reg') || ('to' == 'masq' ) */
if ((compare_url(response->to->url, urlmap[i].masq_url)==STS_SUCCESS) ||
(compare_url(response->to->url, urlmap[i].reg_url)==STS_SUCCESS)){
type=RESTYP_OUTGOING;
DEBUGC(DBCLASS_PROXY,"outgoing response for %s@%s from inbound",
response->from->url->username ?
response->from->url->username : "******",
response->from->url->host ?
response->from->url->host : "*NULL*");
break;
}
}
#else
type = 0;
/*
* did I receive the telegram from a REGISTERED host?
* -> it must be an OUTGOING response
*/
for (i=0; i<URLMAP_SIZE; i++) {
struct in_addr tmp_addr;
if (urlmap[i].active == 0) continue;
if (get_ip_by_host(urlmap[i].true_url->host, &tmp_addr) == STS_FAILURE) {
DEBUGC(DBCLASS_PROXY, "proxy_response: cannot resolve host [%s]",
urlmap[i].true_url);
} else {
DEBUGC(DBCLASS_PROXY, "proxy_response: reghost:%s ip:%s",
urlmap[i].true_url->host, utils_inet_ntoa(from->sin_addr));
if (memcmp(&tmp_addr, &from->sin_addr, sizeof(tmp_addr)) == 0) {
type=RESTYP_OUTGOING;
break;
}
}
}
/*
* is the telegram directed to an internal registered host?
* -> it must be an INCOMING response
*/
if (type == 0) {
for (i=0; i<URLMAP_SIZE; i++) {
if (urlmap[i].active == 0) continue;
/* incoming response ('from' == 'masq') || ('from' == 'reg') */
if ((compare_url(response->from->url, urlmap[i].reg_url)==STS_SUCCESS) ||
(compare_url(response->from->url, urlmap[i].masq_url)==STS_SUCCESS)) {
type=RESTYP_INCOMING;
break;
}
}
}
/* &&&& Open Issue &&&&
it has been seen with cross-provider calls that the FROM may be 'garbled'
(e.g [email protected] for calls made sipphone -> FWD)
How can we deal with this? Should I take into consideration the 'Via'
headers? This is the only clue I have, pointing to the *real* UA.
Maybe I should put in a 'siproxd' ftag value to recognize it as a header
inserted by myself
*/
if ((type == 0) && (!osip_list_eol(response->vias, 0))) {
osip_via_t *via;
struct in_addr addr_via, addr_myself;
int port_via, port_ua;
/* get the via address */
via = (osip_via_t *) osip_list_get (response->vias, 0);
DEBUGC(DBCLASS_PROXY, "proxy_response: check via [%s] for "
"registered UA",via->host);
sts=get_ip_by_host(via->host, &addr_via);
if (sts == STS_FAILURE) {
DEBUGC(DBCLASS_DNS, "proxy_response: cannot resolve VIA [%s]",
via->host);
} else {
for (i=0; i<URLMAP_SIZE; i++) {
if (urlmap[i].active == 0) continue;
/* incoming response (1st via in list points to a registered UA) */
sts=get_ip_by_host(urlmap[i].true_url->host, &addr_myself);
if (sts == STS_FAILURE) {
DEBUGC(DBCLASS_DNS, "proxy_response: cannot resolve "
"true_url [%s]", via->host);
continue;
}
port_via=0;
if (via->port) port_via=atoi(via->port);
if (port_via <= 0) port_via=SIP_PORT;
port_ua=0;
if (urlmap[i].true_url->port)
port_ua=atoi(urlmap[i].true_url->port);
if (port_ua <= 0) port_ua=SIP_PORT;
DEBUGC(DBCLASS_BABBLE, "proxy_response: checking for registered "
"host [%s:%i] <-> [%s:%i]",
urlmap[i].true_url->host, port_ua,
via->host, port_via);
if ((memcmp(&addr_myself, &addr_via, sizeof(addr_myself))==0) &&
(port_via == port_ua)) {
type=RESTYP_INCOMING;
break;
}
}
}
}
#endif
ticket->direction=type;
/*
* ok, we got a response that we are allowed to process.
*/
switch (type) {
/*
* from an external host to the internal masqueraded host
*/
case RESTYP_INCOMING:
DEBUGC(DBCLASS_PROXY,"incoming response for %s@%s from outbound",
response->from->url->username? response->from->url->username:"******",
response->from->url->host? response->from->url->host : "*NULL*");
/*
* Response for INVITE - deal with RTP data in body and
* start RTP proxy stream(s). In case
* of a negative answer, stop RTP stream
*/
if (MSG_IS_RESPONSE_FOR(response,"INVITE")) {
/* positive response, start RTP stream */
if ((MSG_IS_STATUS_1XX(response)) ||
(MSG_IS_STATUS_2XX(response))) {
if (configuration.rtp_proxy_enable == 1) {
sts = proxy_rewrite_invitation_body(response, DIR_INCOMING);
}
/* negative - stop a possibly started RTP stream */
} else if ((MSG_IS_STATUS_4XX(response)) ||
(MSG_IS_STATUS_5XX(response)) ||
(MSG_IS_STATUS_6XX(response))) {
rtp_stop_fwd(osip_message_get_call_id(response), DIR_INCOMING);
rtp_stop_fwd(osip_message_get_call_id(response), DIR_OUTGOING);
}
} /* if INVITE */
/*
* Response for REGISTER - special handling of Contact header
*/
if (MSG_IS_RESPONSE_FOR(response,"REGISTER")) {
/*
* REGISTER returns *my* Contact header information.
* Rewrite Contact header back to represent the true address.
* Other responses do return the Contact header of the sender.
*/
sip_rewrite_contact(ticket, DIR_INCOMING);
}
/*
* Response for SUBSCRIBE
*
* HACK for Grandstream SIP phones (with newer firmware like 1.0.4.40):
* They send a SUBSCRIBE request to the registration server. In
* case of beeing registering directly to siproxd, this request of
* course will eventually be forwarded back to the same UA.
* Grandstream then does reply with an '202' response (A 202
* response merely indicates that the subscription has been
* understood, and that authorization may or may not have been
* granted), which then of course is forwarded back to the phone.
* Ans it seems that the Grandstream can *not* *handle* this
* response, as it immediately sends another SUBSCRIBE request.
* And this games goes on and on and on...
*
* As a workaround we will transform any 202 response to a
* '404 unknown destination'
*
*/
{
osip_header_t *ua_hdr=NULL;
osip_message_get_user_agent(response, 0, &ua_hdr);
if (ua_hdr && ua_hdr->hvalue &&
(osip_strncasecmp(ua_hdr->hvalue,"grandstream", 11)==0) &&
(MSG_IS_RESPONSE_FOR(response,"SUBSCRIBE")) &&
(MSG_TEST_CODE(response, 202))) {
DEBUGC(DBCLASS_PROXY, "proxy_response: Grandstream hack 202->404");
response->status_code=404;
}
}
break;
/*
* from the internal masqueraded host to an external host
*/
case RESTYP_OUTGOING:
DEBUGC(DBCLASS_PROXY,"outgoing response for %s@%s from inbound",
response->from->url->username ?
response->from->url->username : "******",
response->from->url->host ?
response->from->url->host : "*NULL*");
/* rewrite Contact header to represent the masqued address */
sip_rewrite_contact(ticket, DIR_OUTGOING);
/*
* If an 2xx OK or 1xx response, answer to an INVITE request,
* rewrite body
*
* In case of a negative answer, stop RTP stream
*/
if (MSG_IS_RESPONSE_FOR(response,"INVITE")) {
/* positive response, start RTP stream */
if ((MSG_IS_STATUS_1XX(response)) ||
(MSG_IS_STATUS_2XX(response))) {
/* This is an outgoing response, therefore an outgoing stream */
sts = proxy_rewrite_invitation_body(response, DIR_OUTGOING);
/* megative - stop a possibly started RTP stream */
} else if ((MSG_IS_STATUS_4XX(response)) ||
(MSG_IS_STATUS_5XX(response)) ||
(MSG_IS_STATUS_6XX(response))) {
rtp_stop_fwd(osip_message_get_call_id(response), DIR_INCOMING);
rtp_stop_fwd(osip_message_get_call_id(response), DIR_OUTGOING);
}
} /* if INVITE */
break;
default:
DEBUGC(DBCLASS_PROXY, "response from/to unregistered UA (%s@%s)",
response->from->url->username? response->from->url->username:"******",
response->from->url->host? response->from->url->host : "*NULL*");
return STS_FAILURE;
}
/*
* for ALL incoming response include my Record-Route header.
* The local UA will probably send its answer to the topmost
* Route Header (8.1.2 of RFC3261)
*/
if (type == RESTYP_INCOMING) {
DEBUGC(DBCLASS_PROXY,"Adding my Record-Route");
route_add_recordroute(ticket);
} else {
/*
* outgoing packets must not have my record route header, as
* this likely will contain a private IP address (my inbound).
*/
DEBUGC(DBCLASS_PROXY,"Purging Record-Routes (outgoing packet)");
route_purge_recordroute(ticket);
}
/*
* Determine Next-Hop Address
*/
/*&&&& priority probably should be:
* 1) Route header
* 2) fixed outbound proxy
* 3) SIP URI
*/
/*
* check if we need to send to an outbound proxy
*/
if ((type == RESTYP_OUTGOING) &&
(sip_find_outbound_proxy(ticket, &sendto_addr, &port) == STS_SUCCESS)) {
DEBUGC(DBCLASS_PROXY, "proxy_response: have outbound proxy %s:%i",
utils_inet_ntoa(sendto_addr), port);
/*
* Route present?
* If so, fetch address from topmost Route: header and remove it.
*/
} else if ((type == RESTYP_OUTGOING) &&
(response->routes && !osip_list_eol(response->routes, 0))) {
sts=route_determine_nexthop(ticket, &sendto_addr, &port);
if (sts == STS_FAILURE) {
DEBUGC(DBCLASS_PROXY, "proxy_response: route_determine_nexthop failed");
return STS_FAILURE;
}
DEBUGC(DBCLASS_PROXY, "proxy_response: have Route header to %s:%i",
utils_inet_ntoa(sendto_addr), port);
} else {
/* get target address and port from VIA header */
via = (osip_via_t *) osip_list_get (response->vias, 0);
if (via == NULL) {
ERROR("proxy_response: list_get via failed");
return STS_FAILURE;
}
sts = get_ip_by_host(via->host, &sendto_addr);
if (sts == STS_FAILURE) {
DEBUGC(DBCLASS_PROXY, "proxy_response: cannot resolve VIA [%s]",
via->host);
return STS_FAILURE;
}
if (via->port) {
port=atoi(via->port);
} else {
port=SIP_PORT;
}
}
sts = sip_message_to_str(response, &buffer, &buflen);
if (sts != 0) {
ERROR("proxy_response: sip_message_to_str failed");
return STS_FAILURE;
}
sipsock_send(sendto_addr, port, ticket->protocol,
buffer, buflen);
osip_free (buffer);
return STS_SUCCESS;
}
char *
generating_sdp_answer(osip_message_t *request, osip_negotiation_ctx_t *context)
{
sdp_message_t *remote_sdp;
sdp_message_t *local_sdp = NULL;
int i;
char *local_body;
if (context==NULL)
return NULL;
local_body = NULL;
if (MSG_IS_INVITE(request)||MSG_IS_OPTIONS(request)||MSG_IS_RESPONSE_FOR(request, "INVITE"))
{
osip_body_t *body;
body = (osip_body_t *)osip_list_get(&request->bodies,0);
if(body == NULL)
return NULL;
/* remote_sdp = (sdp_message_t *) osip_malloc(sizeof(sdp_message_t)); */
i = sdp_message_init(&remote_sdp);
if (i!=0) return NULL;
/* WE ASSUME IT IS A SDP BODY AND THAT */
/* IT IS THE ONLY ONE, OF COURSE, THIS IS */
/* NOT TRUE */
i = sdp_message_parse(remote_sdp,body->body);
if (i!=0) return NULL;
i = osip_negotiation_ctx_set_remote_sdp(context, remote_sdp);
i = osip_negotiation_ctx_execute_negotiation(eXosip.osip_negotiation, context);
if (i==200)
{
local_sdp = osip_negotiation_ctx_get_local_sdp(context);
i = sdp_message_to_str(local_sdp, &local_body);
remote_sdp = osip_negotiation_ctx_get_remote_sdp(context);
sdp_message_free(remote_sdp);
osip_negotiation_ctx_set_remote_sdp(context, NULL);
if (i!=0) {
OSIP_TRACE(osip_trace(__FILE__,__LINE__,OSIP_ERROR,NULL,"ERROR: Could not parse local SDP answer %i\n",i));
return NULL;
}
return local_body;
}
else if (i==415)
{
OSIP_TRACE(osip_trace(__FILE__,__LINE__,OSIP_INFO1,NULL,"WARNING: Unsupported media %i\n",i));
}
else
{
OSIP_TRACE(osip_trace(__FILE__,__LINE__,OSIP_ERROR,NULL,"ERROR: while building answer to SDP (%i)\n",i));
}
remote_sdp = osip_negotiation_ctx_get_remote_sdp(context);
sdp_message_free(remote_sdp);
osip_negotiation_ctx_set_remote_sdp(context, NULL);
}
return NULL;
}
static struct session *sip_callback(struct session *sip, u_char *data, uint32_t len)
{
osip_message_t *msg;
struct tuple4 addr;
struct session *start;
struct session *rtp;
osip_call_id_t *call_id;
osip_message_init(&msg);
if (!osip_message_parse(msg, (char *)data, len)) {
if (NULL == sip->u.sip_params.call_id) {
/*
* If the session object was created by udp_callback
* we need to fill in the call_id field here because
* udp_callback doesn't know anything about SIP
*/
if (NULL != (call_id = osip_message_get_call_id(msg)))
osip_call_id_clone(call_id, &sip->u.sip_params.call_id);
} else {
/*
* Otherwise check if the session object passed to this
* function call was really the one corresponding to the
* call ID in the SIP packet, in case several SIP calls
* are passed upon the same transport layer protocol,
* source and destination IPv4 address & port combination.
* udp_callback has no way of knowing how to distinguish
* SIP session objects based on call ID, so we have to do
* it here. We just continue searching in the list of all
* tracked sessions for similar SIP objects until we find
* one that has the same call ID, or else we create a new
* SIP session object that corresponds to the new call.
*/
start = sip;
do {
if (NULL == (call_id = osip_message_get_call_id(msg)))
continue;
if (!osip_call_id_match(sip->u.sip_params.call_id, call_id))
break;
} while ((sip = sessions_find(sip->next, TYPE_SIP, 0, &sip->addr)));
if (NULL == sip) {
if (bonus_time) {
osip_message_free(msg);
return start;
}
sip = sessions_add(start->type, &start->addr, NULL);
if (NULL != (call_id = osip_message_get_call_id(msg)))
osip_call_id_clone(call_id, &sip->u.sip_params.call_id);
}
}
/*
* If the current SIP packet is an INVITE message, store the
* advertised source port and IPv4 address. It is not very
* important, since we can do only with the destination part
* (useful in case the capture missed the INVITE packet), but
* it helps discriminating from unrelated packets.
*
* Unfortunately, some SIP implementations such as the one in
* Audiocodes Mediant 1000 SIP gateways actually use a source
* port different from the one they advertised in the INVITE
* message parameters - how outrageous! - so we have to make
* our sessions search engine ignore the source port part by
* zeroing it :-/
*/
if (MSG_IS_INVITE(msg)) {
if (!bonus_time) {
sip_get_address(msg, &sip->u.sip_params.rtp_addr.saddr, &sip->u.sip_params.rtp_addr.source);
#ifndef USING_NON_STUPID_SIP_IMPLEMENTATIONS
sip->u.sip_params.rtp_addr.source = 0;
#endif
}
} else
if (MSG_TEST_CODE(msg, 200)) {
if (MSG_IS_RESPONSE_FOR(msg, "INVITE")) {
if (!bonus_time && sip_get_address(msg, &sip->u.sip_params.rtp_addr.daddr, &sip->u.sip_params.rtp_addr.dest)) {
sessions_add(TYPE_UDP | TYPE_RTP, &sip->u.sip_params.rtp_addr, sip);
sip->u.sip_params.picked_up = 1;
}
} else
if (MSG_IS_RESPONSE_FOR(msg, "BYE") ||
MSG_IS_RESPONSE_FOR(msg, "CANCEL")) {
start = first_session;
while (NULL != (rtp = sessions_find(start, TYPE_RTP, sip->id, NULL))) {
sessions_del(rtp);
start = rtp->next;
}
/*
* Mark for deletion in 2 seconds, in order to give some
* time to the extra ACK packets that might be exchanged
*/
if (sip->type & TYPE_UDP)
sip->timeout = nids_last_pcap_header->ts.tv_sec + 2;
}
}
}
osip_message_free(msg);
return sip;
}
void CIctCallBack::icb_Rev1xx( int type, CSipTransaction* tr, CSipMessage* sip )
{
CZhDialog* jd;
CZhCall* jc;
#ifndef MINISIZE
CZhSubscribe* js;
CZhNotify* jn;
#endif
CTransactionInfo* jinfo = (CTransactionInfo*) tr->GetInstance();
udp_tl_learn_port_from_via(sip);
if (jinfo == NULL)
return;
jd = jinfo->jd;
jc = jinfo->jc;
#ifndef MINISIZE
jn = jinfo->jn;
js = jinfo->js;
#endif
if ((MSG_IS_RESPONSE_FOR(sip, "INVITE")
#ifndef MINISIZE
|| MSG_IS_RESPONSE_FOR(sip, "SUBSCRIBE")
#endif
) && !MSG_TEST_CODE(sip, 100))
{
int i;
if (jd == NULL) /* This transaction initiate a dialog in the case of
INVITE (else it would be attached to a "jd" element. */
{
/* allocate a jd */
jd = new CZhDialog;
i = jd->InitAsUac(sip);
if (i != 0)
{
return;
}
if (jc != NULL)
{
jc->c_dialogs.Add(jd);
jinfo->jd = jd;
CSipTransactionManager *ptm = CSipTransactionManager::getInstance();
ptm->eXosip_update();
}
#ifndef MINISIZE
else if (js != NULL)
{
js->s_dialogs.Add(jd);
jinfo->jd = jd;
CSipTransactionManager *ptm = CSipTransactionManager::getInstance();
ptm->eXosip_update();
}
else if (jn != NULL)
{
jn->n_dialogs.Add(jd);
jinfo->jd = jd;
CSipTransactionManager *ptm = CSipTransactionManager::getInstance();
ptm->eXosip_update();
}
#endif
else
{
}
tr->SetInstance(jinfo);
}
else
{
if (jd->d_dialog == NULL)
{
}
else if (jd->d_dialog->remote_tag == NULL)
{
jd->d_dialog->UpdateRouteSetAsUac(sip);
jd->d_dialog->UpdateTagAsUac(sip);
}
else
{
CUrlParam* tag;
int i;
i = sip->m_to->GetTag(&tag);
if (tag != NULL &&
tag->gvalue != NULL &&
0 == strcmp(jd->d_dialog->remote_tag,tag->gvalue))
{
/* Update only if it is the same dialog */
jd->d_dialog->UpdateRouteSetAsUac(sip);
}
}
}
if (jd != NULL)
jd->d_STATE = JD_TRYING;
if (jd != NULL &&
MSG_IS_RESPONSE_FOR(sip, "INVITE") &&
sip->m_iStatusCode < 180)
{
//report_call_event(EXOSIP_CALL_PROCEEDING, jc, jd, tr);
}
else if (jd != NULL &&
MSG_IS_RESPONSE_FOR(sip, "INVITE") &&
sip->m_iStatusCode >= 180)
{
//report_call_event(EXOSIP_CALL_RINGING, jc, jd, tr);
}
#ifndef MINISIZE
else if (jd != NULL && MSG_IS_RESPONSE_FOR(sip, "SUBSCRIBE"))
{
}
#endif
if (MSG_TEST_CODE(sip, 180) && jd != NULL)
{
jd->d_STATE = JD_RINGING;
}
else if (MSG_TEST_CODE(sip, 183) && jd != NULL)
{
jd->d_STATE = JD_QUEUED;
}
}
}
