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Merged revisions 112785 via svnmerge from

Browse Source 
https://origsvn.digium.com/svn/asterisk/trunk

................
r112785 | phsultan | 2008-04-04 19:32:46 +0200 (Fri, 04 Apr 2008) | 15 lines

Merged revisions 112766 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines

Prevent call connections when codecs don't match.

(closes issue #10604)
Reported by: keepitcool
Patches:
      branch-1.4-10604-2.diff uploaded by phsultan (license 73)
Tested by: phsultan
........

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@112786 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.6.0
Philippe Sultan 17 years ago
parent c9c67f0685
commit acf6ac5a7f
1 changed files with 124 additions and 45 deletions
Show Stats Download Patch File Download Diff File

169
channels/chan_gtalk.c
Unescape View File

@ -170,6 +170,8 @@ static int gtalk_digit_end(struct ast_channel *ast, char digit, unsigned int dur
static int gtalk_call(struct ast_channel *ast, char *dest, int timeout);
static int gtalk_hangup(struct ast_channel *ast);
static int gtalk_answer(struct ast_channel *ast);
static int gtalk_action(struct gtalk *client, struct gtalk_pvt *p, const char *action);
static void gtalk_free_pvt(struct gtalk *client, struct gtalk_pvt *p);
static int gtalk_newcall(struct gtalk *client, ikspak *pak);
static struct ast_frame *gtalk_read(struct ast_channel *ast);
static int gtalk_write(struct ast_channel *ast, struct ast_frame *f);
@ -273,6 +275,7 @@ static struct gtalk *find_gtalk(char *name, char *connection)
static int add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodecs)
{
int res = 0;
char *format = ast_getformatname(codec);
if (!strcasecmp("ulaw", format)) {
@ -296,6 +299,7 @@ static int add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodec
iks_insert_attrib(payload_eg711u, "bitrate","64000");
iks_insert_node(dcodecs, payload_pcmu);
iks_insert_node(dcodecs, payload_eg711u);
res ++;
}
if (!strcasecmp("alaw", format)) {
iks *payload_eg711a, *payload_pcma;
@ -318,6 +322,7 @@ static int add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodec
iks_insert_attrib(payload_eg711a, "bitrate","64000");
iks_insert_node(dcodecs, payload_pcma);
iks_insert_node(dcodecs, payload_eg711a);
res ++;
}
if (!strcasecmp("ilbc", format)) {
iks *payload_ilbc = iks_new("payload-type");
@ -330,6 +335,7 @@ static int add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodec
iks_insert_attrib(payload_ilbc, "clockrate","8000");
iks_insert_attrib(payload_ilbc, "bitrate","13300");
iks_insert_node(dcodecs, payload_ilbc);
res ++;
}
if (!strcasecmp("g723", format)) {
iks *payload_g723 = iks_new("payload-type");
@ -342,6 +348,7 @@ static int add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodec
iks_insert_attrib(payload_g723, "clockrate","8000");
iks_insert_attrib(payload_g723, "bitrate","6300");
iks_insert_node(dcodecs, payload_g723);
res ++;
}
if (!strcasecmp("speex", format)) {
iks *payload_speex = iks_new("payload-type");
@ -354,9 +361,21 @@ static int add_codec_to_answer(const struct gtalk_pvt *p, int codec, iks *dcodec
iks_insert_attrib(payload_speex, "clockrate","8000");
iks_insert_attrib(payload_speex, "bitrate","11000");
iks_insert_node(dcodecs, payload_speex);
res++;
}
if (!strcasecmp("gsm", format)) {
iks *payload_gsm = iks_new("payload-type");
if(!payload_gsm) {
ast_log(LOG_WARNING,"Failed to allocate iks node");
return -1;
}
iks_insert_attrib(payload_gsm, "id", "103");
iks_insert_attrib(payload_gsm, "name", "gsm");
iks_insert_node(dcodecs, payload_gsm);
res++;
}
ast_rtp_lookup_code(p->rtp, 1, codec);
return 0;
return res;
}
static int gtalk_invite(struct gtalk_pvt *p, char *to, char *from, char *sid, int initiator)
@ -366,7 +385,7 @@ static int gtalk_invite(struct gtalk_pvt *p, char *to, char *from, char *sid, in
int x;
int pref_codec = 0;
int alreadysent = 0;
int codecs_num = 0;
iq = iks_new("iq");
gtalk = iks_new("session");
@ -393,14 +412,16 @@ static int gtalk_invite(struct gtalk_pvt *p, char *to, char *from, char *sid, in
continue;
if (alreadysent & pref_codec)
continue;
add_codec_to_answer(p, pref_codec, dcodecs);
codecs_num = add_codec_to_answer(p, pref_codec, dcodecs);
alreadysent |= pref_codec;
}
iks_insert_attrib(payload_telephone, "id", "106");
iks_insert_attrib(payload_telephone, "name", "telephone-event");
iks_insert_attrib(payload_telephone, "clockrate", "8000");
if (codecs_num) {
/* only propose DTMF within an audio session */
iks_insert_attrib(payload_telephone, "id", "106");
iks_insert_attrib(payload_telephone, "name", "telephone-event");
iks_insert_attrib(payload_telephone, "clockrate", "8000");
}
iks_insert_attrib(transport,"xmlns","http://www.google.com/transport/p2p");
iks_insert_attrib(iq, "type", "set");
@ -568,12 +589,41 @@ static int gtalk_is_answered(struct gtalk *client, ikspak *pak)
{
struct gtalk_pvt *tmp;
char *from;
ast_debug(1, "The client is %s\n", client->name);
iks *codec;
char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ];
int peernoncodeccapability;
ast_log(LOG_DEBUG, "The client is %s\n", client->name);
/* Make sure our new call doesn't exist yet */
for (tmp = client->p; tmp; tmp = tmp->next) {
if (iks_find_with_attrib(pak->x, "session", "id", tmp->sid))
break;
}
/* codec points to the first <payload-type/> tag */
codec = iks_child(iks_child(iks_child(pak->x)));
while (codec) {
ast_rtp_set_m_type(tmp->rtp, atoi(iks_find_attrib(codec, "id")));
ast_rtp_set_rtpmap_type(tmp->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next(codec);
}
/* Now gather all of the codecs that we are asked for */
ast_rtp_get_current_formats(tmp->rtp, &tmp->peercapability, &peernoncodeccapability);
/* at this point, we received an awser from the remote Gtalk client,
which allows us to compare capabilities */
tmp->jointcapability = tmp->capability & tmp->peercapability;
if (!tmp->jointcapability) {
ast_log(LOG_WARNING, "Capabilities don't match : us - %s, peer - %s, combined - %s \n", ast_getformatname_multiple(s1, BUFSIZ, tmp->capability),
ast_getformatname_multiple(s2, BUFSIZ, tmp->peercapability),
ast_getformatname_multiple(s3, BUFSIZ, tmp->jointcapability));
/* close session if capabilities don't match */
ast_queue_hangup(tmp->owner);
return -1;
}
from = iks_find_attrib(pak->x, "to");
if(!from)
@ -875,7 +925,16 @@ static struct gtalk_pvt *gtalk_alloc(struct gtalk *client, const char *us, const
ast_copy_string(tmp->us, us, sizeof(tmp->us));
tmp->initiator = 1;
}
/* clear codecs */
tmp->rtp = ast_rtp_new_with_bindaddr(sched, io, 1, 0, bindaddr.sin_addr);
ast_rtp_pt_clear(tmp->rtp);
/* add user configured codec capabilites */
if (client->capability)
tmp->capability = client->capability;
else if (global_capability)
tmp->capability = global_capability;
tmp->parent = client;
if (!tmp->rtp) {
ast_log(LOG_WARNING, "Out of RTP sessions?\n");
@ -921,7 +980,6 @@ static struct ast_channel *gtalk_new(struct gtalk *client, struct gtalk_pvt *i,
/* Select our native format based on codec preference until we receive
something from another device to the contrary. */
/* ast_verbose("XXXXXXXXXXXXX\nXXX i->jointcapability = %X\nXXX i->capability = %X\nXXX global_capability %X\n XXXXXXXXXXXX\n",i->jointcapability,i->capability,global_capability); */
if (i->jointcapability)
what = i->jointcapability;
else if (i->capability)
@ -1064,6 +1122,9 @@ static int gtalk_newcall(struct gtalk *client, ikspak *pak)
int res;
iks *codec;
char *from = NULL;
char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ];
int peernoncodeccapability;
/* Make sure our new call doesn't exist yet */
from = iks_find_attrib(pak->x,"to");
if(!from)
@ -1083,46 +1144,65 @@ static int gtalk_newcall(struct gtalk *client, ikspak *pak)
ast_log(LOG_WARNING, "Unable to allocate gtalk structure!\n");
return -1;
}
chan = gtalk_new(client, p, AST_STATE_DOWN, pak->from->user);
if (chan) {
ast_mutex_lock(&p->lock);
ast_copy_string(p->them, pak->from->full, sizeof(p->them));
if (iks_find_attrib(pak->query, "id")) {
ast_copy_string(p->sid, iks_find_attrib(pak->query, "id"),
sizeof(p->sid));
}
if (!chan) {
gtalk_free_pvt(client, p);
return -1;
}
codec = iks_child(iks_child(iks_child(pak->x)));
while (codec) {
ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio",
iks_find_attrib(codec, "name"), 0);
codec = iks_next(codec);
}
ast_mutex_lock(&p->lock);
ast_copy_string(p->them, pak->from->full, sizeof(p->them));
if (iks_find_attrib(pak->query, "id")) {
ast_copy_string(p->sid, iks_find_attrib(pak->query, "id"),
sizeof(p->sid));
}
/* codec points to the first <payload-type/> tag */
codec = iks_child(iks_child(iks_child(pak->x)));
while (codec) {
ast_rtp_set_m_type(p->rtp, atoi(iks_find_attrib(codec, "id")));
ast_rtp_set_rtpmap_type(p->rtp, atoi(iks_find_attrib(codec, "id")), "audio", iks_find_attrib(codec, "name"), 0);
codec = iks_next(codec);
}
/* Now gather all of the codecs that we are asked for */
ast_rtp_get_current_formats(p->rtp, &p->peercapability, &peernoncodeccapability);
p->jointcapability = p->capability & p->peercapability;
ast_mutex_unlock(&p->lock);
ast_mutex_unlock(&p->lock);
ast_setstate(chan, AST_STATE_RING);
res = ast_pbx_start(chan);
ast_setstate(chan, AST_STATE_RING);
if (!p->jointcapability) {
ast_log(LOG_WARNING, "Capabilities don't match : us - %s, peer - %s, combined - %s \n", ast_getformatname_multiple(s1, BUFSIZ, p->capability),
ast_getformatname_multiple(s2, BUFSIZ, p->peercapability),
ast_getformatname_multiple(s3, BUFSIZ, p->jointcapability));
/* close session if capabilities don't match */
gtalk_action(client, p, "reject");
p->alreadygone = 1;
gtalk_hangup(chan);
return -1;
}
switch (res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
gtalk_response(client, from, pak, "service-unavailable", NULL);
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
gtalk_response(client, from, pak, "service-unavailable", NULL);
break;
case AST_PBX_SUCCESS:
gtalk_response(client, from, pak, NULL, NULL);
gtalk_invite_response(p, p->them, p->us,p->sid, 0);
gtalk_create_candidates(client, p, p->sid, p->them, p->us);
/* nothing to do */
break;
}
} else {
gtalk_free_pvt(client, p);
res = ast_pbx_start(chan);
switch (res) {
case AST_PBX_FAILED:
ast_log(LOG_WARNING, "Failed to start PBX :(\n");
gtalk_response(client, from, pak, "service-unavailable", NULL);
break;
case AST_PBX_CALL_LIMIT:
ast_log(LOG_WARNING, "Failed to start PBX (call limit reached) \n");
gtalk_response(client, from, pak, "service-unavailable", NULL);
break;
case AST_PBX_SUCCESS:
gtalk_response(client, from, pak, NULL, NULL);
gtalk_invite_response(p, p->them, p->us,p->sid, 0);
gtalk_create_candidates(client, p, p->sid, p->them, p->us);
/* nothing to do */
break;
}
return 1;
}
@ -1465,7 +1545,6 @@ static int gtalk_call(struct ast_channel *ast, char *dest, int timeout)
}
ast_setstate(ast, AST_STATE_RING);
p->jointcapability = p->capability;
if (!p->ringrule) {
ast_copy_string(p->ring, p->parent->connection->mid, sizeof(p->ring));
p->ringrule = iks_filter_add_rule(p->parent->connection->f, gtalk_ringing_ack, p,

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