summaryrefslogtreecommitdiff
diff options
context:
space:
mode:
authorRussell Bryant <russell@russellbryant.com>2010-03-05 05:03:41 +0000
committerRussell Bryant <russell@russellbryant.com>2010-03-05 05:03:41 +0000
commit7f8e8d01de3c0c76c889d0fc08abf5f56e0d1ecd (patch)
tree7214c674da905296971cf6672684b3d422569a92
parent6d166a9af9a2ded74f8830daf485d430a88e9ff0 (diff)
Fix up some of chan_sip's usage of the RTP engine API.
The get_local_address() function for an RTP instance was used when building an SDP, but the results were not honored. The RTP engine activate() function was not being used once we have determined that media will now flow. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250917 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r--channels/chan_sip.c46
1 files changed, 43 insertions, 3 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index cb7fba9d7..793a31b3c 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -8791,7 +8791,19 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
dest->sin_port = p->redirip.sin_port;
dest->sin_addr = p->redirip.sin_addr;
} else {
- dest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr : p->ourip.sin_addr;
+ /*
+ * Audio Destination IP:
+ *
+ * 1. Specifically configured media address.
+ * 2. Local address as specified by the RTP engine.
+ * 3. The local IP as defined by chan_sip.
+ *
+ * Audio Destination Port:
+ *
+ * 1. Provided by the RTP engine.
+ */
+ dest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr :
+ (sin->sin_addr.s_addr ? sin->sin_addr : p->ourip.sin_addr);
dest->sin_port = sin->sin_port;
}
if (needvideo) {
@@ -8800,7 +8812,19 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
vdest->sin_addr = p->vredirip.sin_addr;
vdest->sin_port = p->vredirip.sin_port;
} else {
- vdest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr : p->ourip.sin_addr;
+ /*
+ * Video Destination IP:
+ *
+ * 1. Specifically configured media address.
+ * 2. Local address as specified by the RTP engine.
+ * 3. The local IP as defined by chan_sip.
+ *
+ * Video Destination Port:
+ *
+ * 1. Provided by the RTP engine.
+ */
+ vdest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr :
+ (vsin->sin_addr.s_addr ? vsin->sin_addr : p->ourip.sin_addr);
vdest->sin_port = vsin->sin_port;
}
}
@@ -8810,7 +8834,19 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, int needtext
tdest->sin_addr = p->tredirip.sin_addr;
tdest->sin_port = p->tredirip.sin_port;
} else {
- tdest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr : p->ourip.sin_addr;
+ /*
+ * Text Destination IP:
+ *
+ * 1. Specifically configured media address.
+ * 2. Local address as specified by the RTP engine.
+ * 3. The local IP as defined by chan_sip.
+ *
+ * Text Destination Port:
+ *
+ * 1. Provided by the RTP engine.
+ */
+ tdest->sin_addr = media_address.sin_addr.s_addr ? media_address.sin_addr :
+ (tsin->sin_addr.s_addr ? tsin->sin_addr : p->ourip.sin_addr);
tdest->sin_port = tsin->sin_port;
}
}
@@ -9235,6 +9271,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const
ast_debug(1, "Setting framing from config on incoming call\n");
ast_rtp_codecs_packetization_set(ast_rtp_instance_get_codecs(p->rtp), p->rtp, &p->prefs);
}
+ ast_rtp_instance_activate(p->rtp);
try_suggested_sip_codec(p);
if (p->t38.state == T38_ENABLED) {
add_sdp(&resp, p, oldsdp, TRUE, TRUE);
@@ -16671,6 +16708,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
/* Queue a progress frame only if we have SDP in 180 or 182 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
+ ast_rtp_instance_activate(p->rtp);
}
check_pendings(p);
break;
@@ -16708,6 +16746,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
/* Queue a progress frame */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
+ ast_rtp_instance_activate(p->rtp);
} else {
/* Alcatel PBXs are known to send 183s with no SDP after sending
* a 100 Trying response. We're just going to treat this sort of thing
@@ -16730,6 +16769,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest
/* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
/* For re-invites, we try to recover */
ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
+ ast_rtp_instance_activate(p->rtp);
}
if (!req->ignore && p->owner && (get_rpid(p, req) || !reinvite)) {