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authorKevin Harwell <kharwell@digium.com>2014-12-09 20:02:29 +0000
committerKevin Harwell <kharwell@digium.com>2014-12-09 20:02:29 +0000
commit525c823b4b7e63dc74669ebb5284397713804c75 (patch)
tree6d4736bbfe744229826b80f6f567bd8e0a0547d3 /channels
parent664067e3189b29bd00276725d2293363c9b9ebe1 (diff)
Direct Media calls within private network sometimes get one way audio
When endpoints with direct_media enabled, behind a firewall (Asterisk on a separate network) and were bridged sometimes Asterisk would send the ip address of the firewall in the sdp to one of the phones in the reinvite resulting in one way audio. When sending the reinvite Asterisk will retrieve the media address from the associated rtp instance, but if frames were being read this can be overwritten with another address (in this case the firewall's). This patch ensures that Asterisk uses the original device address when using direct media. ASTERISK-24563 Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4216/ ........ Merged revisions 429195 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429196 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c4
1 files changed, 2 insertions, 2 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index ab03580ad..ee55fd38d 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -23062,7 +23062,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
} else if (!reinvite) {
struct ast_sockaddr remote_address = {{0,}};
- ast_rtp_instance_get_remote_address(p->rtp, &remote_address);
+ ast_rtp_instance_get_requested_target_address(p->rtp, &remote_address);
if (ast_sockaddr_isnull(&remote_address) || (!ast_strlen_zero(p->theirprovtag) && strcmp(p->theirtag, p->theirprovtag))) {
ast_log(LOG_WARNING, "Received response: \"200 OK\" from '%s' without SDP\n", p->relatedpeer->name);
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
@@ -32291,7 +32291,7 @@ static int sip_allow_anyrtp_remote(struct ast_channel *chan1, struct ast_rtp_ins
if (ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
struct ast_sockaddr us = { { 0, }, }, them = { { 0, }, };
- ast_rtp_instance_get_remote_address(instance, &them);
+ ast_rtp_instance_get_requested_target_address(instance, &them);
ast_rtp_instance_get_local_address(instance, &us);
if (ast_apply_acl(acl, &them, "SIP Direct Media ACL: ") == AST_SENSE_DENY) {