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Add support for applying direct media ACLs between differing channel technologies.

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Review: https://reviewboard.asterisk.org/r/2122/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374414 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2012-10-04 13:49:45 +00:00
parent 4af961a03a
commit d78f7f92b2
3 changed files with 44 additions and 79 deletions
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91
channels/chan_sip.c
View File

@@ -31867,25 +31867,6 @@ static int reload_config(enum channelreloadreason reason)
return 0;
}
static int apply_directmedia_acl(struct sip_pvt *p, struct ast_acl_list *directmediaacl, const char *op)
{
struct ast_sockaddr us = { { 0, }, }, them = { { 0, }, };
int res = AST_SENSE_ALLOW;
ast_rtp_instance_get_remote_address(p->rtp, &them);
ast_rtp_instance_get_local_address(p->rtp, &us);
if ((res = ast_apply_acl(directmediaacl, &them, "SIP Direct Media ACL: ")) == AST_SENSE_DENY) {
const char *us_addr = ast_strdupa(ast_sockaddr_stringify(&us));
const char *them_addr = ast_strdupa(ast_sockaddr_stringify(&them));
ast_debug(3, "Reinvite %s to %s denied by directmedia ACL on %s\n",
op, them_addr, us_addr);
}
return res;
}
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
{
struct sip_pvt *p;
@@ -31947,72 +31928,56 @@ 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)
static int sip_allow_anyrtp_remote(struct ast_channel *chan1, struct ast_rtp_instance *instance, const char *rtptype)
{
struct sip_pvt *p1 = NULL, *p2 = NULL;
struct ast_acl_list *p2_directmediaacl = NULL; /* opposed directmediaha for comparing against first channel host address */
struct ast_acl_list *p1_directmediaacl = NULL; /* opposed directmediaha for comparing against second channel host address */
struct sip_pvt *p;
struct ast_acl_list *acl = NULL;
int res = 1;
if (!(p1 = ast_channel_tech_pvt(chan1))) {
if (!(p = ast_channel_tech_pvt(chan1))) {
return 0;
}
if (!(p2 = ast_channel_tech_pvt(chan2))) {
return 0;
sip_pvt_lock(p);
if (p->relatedpeer && p->relatedpeer->directmediaacl) {
acl = ast_duplicate_acl_list(p->relatedpeer->directmediaacl);
}
sip_pvt_unlock(p);
if (!acl) {
return res;
}
sip_pvt_lock(p2);
if (p2->relatedpeer && p2->relatedpeer->directmediaacl) {
p2_directmediaacl = ast_duplicate_acl_list(p2->relatedpeer->directmediaacl);
}
sip_pvt_unlock(p2);
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
struct ast_sockaddr us = { { 0, }, }, them = { { 0, }, };
sip_pvt_lock(p1);
if (p1->relatedpeer && p1->relatedpeer->directmediaacl) {
p1_directmediaacl = ast_duplicate_acl_list(p1->relatedpeer->directmediaacl);
}
ast_rtp_instance_get_remote_address(instance, &them);
ast_rtp_instance_get_local_address(instance, &us);
if (ast_apply_acl(acl, &them, "SIP Direct Media ACL: ") == AST_SENSE_DENY) {
const char *us_addr = ast_strdupa(ast_sockaddr_stringify(&us));
const char *them_addr = ast_strdupa(ast_sockaddr_stringify(&them));
ast_debug(3, "Reinvite %s to %s denied by directmedia ACL on %s\n",
rtptype, them_addr, us_addr);
if (p2_directmediaacl && ast_test_flag(&p1->flags[0], SIP_DIRECT_MEDIA)) {
if (!apply_directmedia_acl(p1, p2_directmediaacl, rtptype)) {
res = 0;
}
}
sip_pvt_unlock(p1);
if (res == 0) {
goto allow_anyrtp_remote_end;
}
sip_pvt_lock(p2);
if (p1_directmediaacl && ast_test_flag(&p2->flags[0], SIP_DIRECT_MEDIA)) {
if (!apply_directmedia_acl(p2, p1_directmediaacl, rtptype)) {
res = 0;
}
}
sip_pvt_unlock(p2);
allow_anyrtp_remote_end:
if (p2_directmediaacl) {
p2_directmediaacl = ast_free_acl_list(p2_directmediaacl);
}
if (p1_directmediaacl) {
p1_directmediaacl = ast_free_acl_list(p1_directmediaacl);
}
ast_free_acl_list(acl);
return res;
}
static int sip_allow_rtp_remote(struct ast_channel *chan1, struct ast_channel *chan2)
static int sip_allow_rtp_remote(struct ast_channel *chan1, struct ast_rtp_instance *instance)
{
return sip_allow_anyrtp_remote(chan1, chan2, "audio");
return sip_allow_anyrtp_remote(chan1, instance, "audio");
}
static int sip_allow_vrtp_remote(struct ast_channel *chan1, struct ast_channel *chan2)
static int sip_allow_vrtp_remote(struct ast_channel *chan1, struct ast_rtp_instance *instance)
{
return sip_allow_anyrtp_remote(chan1, chan2, "video");
return sip_allow_anyrtp_remote(chan1, instance, "video");
}
static enum ast_rtp_glue_result sip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)

6
include/asterisk/rtp_engine.h
View File

@@ -509,10 +509,9 @@ struct ast_rtp_glue {
/*!
* \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);
int (*allow_rtp_remote)(struct ast_channel *chan1, struct ast_rtp_instance *instance);
/*!
* \brief Callback for retrieving the RTP instance carrying video
* \note This function increases the reference count on the returned RTP instance.
@@ -521,10 +520,9 @@ struct ast_rtp_glue {
/*!
* \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);
int (*allow_vrtp_remote)(struct ast_channel *chan1, struct ast_rtp_instance *instance);
/*!
* \brief Callback for retrieving the RTP instance carrying text

12
main/rtp_engine.c
View File

@@ -1484,19 +1484,21 @@ 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)) {
/* Apply any limitations on direct media bridging that may be present */
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 (glue0->allow_rtp_remote && !(glue0->allow_rtp_remote(c0, instance1))) {
/* 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;
} else if (glue1->allow_rtp_remote && !(glue1->allow_rtp_remote(c1, instance0))) {
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 (glue0->allow_vrtp_remote && !(glue0->allow_vrtp_remote(c0, instance1))) {
/* 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;
}
} else if (glue1->allow_vrtp_remote && !(glue1->allow_vrtp_remote(c1, instance0))) {
video_glue0_res = video_glue1_res = AST_RTP_GLUE_RESULT_LOCAL;
}
}