[asterisk-commits] phsultan: trunk r112785 - in /trunk: ./ channels/chan_gtalk.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Fri Apr 4 12:32:46 CDT 2008
Author: phsultan
Date: Fri Apr 4 12:32:46 2008
New Revision: 112785
URL: http://svn.digium.com/view/asterisk?view=rev&rev=112785
Log:
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
........
Modified:
trunk/ (props changed)
trunk/channels/chan_gtalk.c
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_gtalk.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_gtalk.c?view=diff&rev=112785&r1=112784&r2=112785
==============================================================================
--- trunk/channels/chan_gtalk.c (original)
+++ trunk/channels/chan_gtalk.c Fri Apr 4 12:32:46 2008
@@ -170,6 +170,8 @@
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 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
{
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 @@
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 @@
/* 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 @@
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 @@
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));
- }
-
- 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);
- }
+ if (!chan) {
+ gtalk_free_pvt(client, p);
+ return -1;
+ }
+
+ 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);
-
- 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);
- }
+ 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;
+ }
+
+ 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 @@
}
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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