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chan_sip: fix problem directmediapermit/deny uses the wrong address

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When remotely bridging calls with directmedia, Asterisk would check
the address of the peers/users holding directmedia ACLs (set via
directmediapermit/directmediadeny) instead of the bridged peer. This
is similar to r366547, but trunk specific and involves changes to
the rtpengine instead of just chan_sip.

(closes issue AST-876)
review: https://reviewboard.asterisk.org/r/1924/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@367640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
certified/11.2
Jonathan Rose 13 years ago
parent d0ed332750
commit bdaecbb66b
3 changed files with 105 additions and 15 deletions
Show Stats Download Patch File Download Diff File

89
channels/chan_sip.c
Unescape View File

@ -30370,7 +30370,7 @@ static int reload_config(enum channelreloadreason reason)
return 0;
}
static int apply_directmedia_ha(struct sip_pvt *p, const char *op)
static int apply_directmedia_ha(struct sip_pvt *p, struct ast_ha *directmediaha, const char *op)
{
struct ast_sockaddr us = { { 0, }, }, them = { { 0, }, };
int res = AST_SENSE_ALLOW;
@ -30378,7 +30378,7 @@ static int apply_directmedia_ha(struct sip_pvt *p, const char *op)
ast_rtp_instance_get_remote_address(p->rtp, &them);
ast_rtp_instance_get_local_address(p->rtp, &us);
if ((res = ast_apply_ha(p->directmediaha, &them)) == AST_SENSE_DENY) {
if ((res = ast_apply_ha(directmediaha, &them)) == AST_SENSE_DENY) {
const char *us_addr = ast_strdupa(ast_sockaddr_stringify(&us));
const char *them_addr = ast_strdupa(ast_sockaddr_stringify(&them));
@ -30398,12 +30398,10 @@ static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
if (!p) {
return NULL;
}
sip_pvt_lock(p);
if (p->udptl && ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
if (apply_directmedia_ha(p, "UDPTL T.38 data")) {
udptl = p->udptl;
}
udptl = p->udptl;
}
sip_pvt_unlock(p);
return udptl;
@ -30452,6 +30450,74 @@ static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
return 0;
}
static int sip_allow_anyrtp_remote(struct ast_channel *chan1, struct ast_channel *chan2, char *rtptype)
{
struct sip_pvt *p1 = NULL, *p2 = NULL;
struct ast_ha *p2_directmediaha = NULL; /* opposed directmediaha for comparing against first channel host address */
struct ast_ha *p1_directmediaha = NULL; /* opposed directmediaha for comparing against second channel host address */
int res = 1;
if (!(p1 = ast_channel_tech_pvt(chan1))) {
return 0;
}
if (!(p2 = ast_channel_tech_pvt(chan2))) {
return 0;
}
sip_pvt_lock(p2);
if (p2->relatedpeer->directmediaha) {
p2_directmediaha = ast_duplicate_ha_list(p2->relatedpeer->directmediaha);
}
sip_pvt_unlock(p2);
sip_pvt_lock(p1);
if (p1->relatedpeer->directmediaha) {
p1_directmediaha = ast_duplicate_ha_list(p1->relatedpeer->directmediaha);
}
if (ast_test_flag(&p1->flags[0], SIP_DIRECT_MEDIA)) {
if (!apply_directmedia_ha(p1, p2_directmediaha, rtptype)) {
res = 0;
}
}
sip_pvt_unlock(p1);
if (res == 0) {
goto allow_anyrtp_remote_end;
}
sip_pvt_lock(p2);
if (ast_test_flag(&p2->flags[0], SIP_DIRECT_MEDIA)) {
if (!apply_directmedia_ha(p2, p1_directmediaha, rtptype)) {
res = 0;
}
}
sip_pvt_unlock(p2);
allow_anyrtp_remote_end:
if (p2_directmediaha) {
ast_free_ha(p2_directmediaha);
}
if (p1_directmediaha) {
ast_free_ha(p1_directmediaha);
}
return res;
}
static int sip_allow_rtp_remote(struct ast_channel *chan1, struct ast_channel *chan2)
{
return sip_allow_anyrtp_remote(chan1, chan2, "audio");
}
static int sip_allow_vrtp_remote(struct ast_channel *chan1, struct ast_channel *chan2)
{
return sip_allow_anyrtp_remote(chan1, chan2, "video");
}
static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct sip_pvt *p = NULL;
@ -30472,9 +30538,6 @@ static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struc
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
if (!apply_directmedia_ha(p, "audio")) {
res = AST_RTP_GLUE_RESULT_FORBID;
}
} else if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA_NAT)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
} else if (ast_test_flag(&global_jbconf, AST_JB_FORCED)) {
@ -30510,9 +30573,6 @@ static enum ast_rtp_glue_result sip_get_vrtp_peer(struct ast_channel *chan, stru
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
if (!apply_directmedia_ha(p, "video")) {
res = AST_RTP_GLUE_RESULT_FORBID;
}
}
sip_pvt_unlock(p);
@ -30540,9 +30600,6 @@ static enum ast_rtp_glue_result sip_get_trtp_peer(struct ast_channel *chan, stru
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
res = AST_RTP_GLUE_RESULT_REMOTE;
if (!apply_directmedia_ha(p, "text")) {
res = AST_RTP_GLUE_RESULT_FORBID;
}
}
sip_pvt_unlock(p);
@ -30680,7 +30737,9 @@ static void sip_get_codec(struct ast_channel *chan, struct ast_format_cap *resul
static struct ast_rtp_glue sip_rtp_glue = {
.type = "SIP",
.get_rtp_info = sip_get_rtp_peer,
.allow_rtp_remote = sip_allow_rtp_remote,
.get_vrtp_info = sip_get_vrtp_peer,
.allow_vrtp_remote = sip_allow_vrtp_remote,
.get_trtp_info = sip_get_trtp_peer,
.update_peer = sip_set_rtp_peer,
.get_codec = sip_get_codec,

15
include/asterisk/rtp_engine.h
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@ -404,11 +404,26 @@ struct ast_rtp_glue {
* \note This function increases the reference count on the returned RTP instance.
*/
enum ast_rtp_glue_result (*get_rtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
/*!
* \brief Used to prevent two channels from remotely bridging audio rtp if the channel tech has a
* reason for prohibiting it based on qualities that need to be compared from both channels.
* \note This function should only be called with two channels of the same technology
* \note This function may be NULL for a given channel driver. This should be accounted for and if that is the case, function this is not used.
*/
int (*allow_rtp_remote)(struct ast_channel *chan1, struct ast_channel *chan2);
/*!
* \brief Callback for retrieving the RTP instance carrying video
* \note This function increases the reference count on the returned RTP instance.
*/
enum ast_rtp_glue_result (*get_vrtp_info)(struct ast_channel *chan, struct ast_rtp_instance **instance);
/*!
* \brief Used to prevent two channels from remotely bridging video rtp if the channel tech has a
* reason for prohibiting it based on qualities that need to be compared from both channels.
* \note This function should only be called with two channels of the same technology
* \note This function may be NULL for a given channel driver. This should be accounted for and if that is the case, this function is not used.
*/
int (*allow_vrtp_remote)(struct ast_channel *chan1, struct ast_channel *chan2);
/*!
* \brief Callback for retrieving the RTP instance carrying text
* \note This function increases the reference count on the returned RTP instance.

16
main/rtp_engine.c
Unescape View File

@ -1339,6 +1339,22 @@ enum ast_bridge_result ast_rtp_instance_bridge(struct ast_channel *c0, struct as
audio_glue1_res = glue1->get_rtp_info(c1, &instance1);
video_glue1_res = glue1->get_vrtp_info ? glue1->get_vrtp_info(c1, &vinstance1) : AST_RTP_GLUE_RESULT_FORBID;
/* If the channels are of the same technology, they might have limitations on remote bridging */
if (ast_channel_tech(c0) == ast_channel_tech(c1)) {
if (audio_glue0_res == audio_glue1_res && audio_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
if (glue0->allow_rtp_remote && !(glue0->allow_rtp_remote(c0, c1))) {
/* If the allow_rtp_remote indicates that remote isn't allowed, revert to local bridge */
audio_glue0_res = audio_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
}
}
if (video_glue0_res == video_glue1_res && video_glue1_res == AST_RTP_GLUE_RESULT_REMOTE) {
if (glue0->allow_vrtp_remote && !(glue0->allow_vrtp_remote(c0, c1))) {
/* if the allow_vrtp_remote indicates that remote isn't allowed, revert to local bridge */
video_glue0_res = video_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
}
}
}
/* If we are carrying video, and both sides are not going to remotely bridge... fail the native bridge */
if (video_glue0_res != AST_RTP_GLUE_RESULT_FORBID && (audio_glue0_res != AST_RTP_GLUE_RESULT_REMOTE || video_glue0_res != AST_RTP_GLUE_RESULT_REMOTE)) {
audio_glue0_res = AST_RTP_GLUE_RESULT_FORBID;

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