Allow for media to arrive from an alternate source when responding to a reinvite with 491.

When we receive a SIP reinvite, it is possible that we may not be able to process the
reinvite immediately since we have also sent a reinvite out ourselves. The problem is
that whoever sent us the reinvite may have also sent a reinvite out to another party,
and that reinvite may have succeeded.

As a result, even though we are not going to accept the reinvite we just received, it
is important for us to not have problems if we suddenly start receiving RTP from a new
source. The fix for this is to grab the media source information from the SDP of the
reinvite that we receive. This information is passed to the RTP layer so that it will
know about the alternate source for media.

Review: https://reviewboard.asterisk.org/r/252



git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@197588 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 72 insertions, 0 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index ce859d7c1..d6c409be8 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5111,6 +5111,63 @@ static void change_hold_state(struct sip_pvt *dialog, struct sip_request *req, i
 	return;
 }
 
+enum media_type {
+	SDP_AUDIO,
+	SDP_VIDEO,
+};
+
+static int get_ip_and_port_from_sdp(struct sip_request *req, const enum media_type media, struct sockaddr_in *sin)
+{
+	const char *m;
+	const char *c;
+	int miterator = req->sdp_start;
+	int citerator = req->sdp_start;
+	int x = 0;
+	int numberofports;
+	int len;
+	char host[258] = ""; /*Initialize to empty so we will know if we have any input */
+	struct ast_hostent audiohp;
+	struct hostent *hp;
+
+	c = get_sdp_iterate(&citerator, req, "c");
+	if (sscanf(c, "IN IP4 %256s", host) != 1) {
+		ast_log(LOG_WARNING, "Invalid host in c= line, '%s'\n", c);
+		/* Continue since there may be a valid host in a c= line specific to the audio stream */
+	}
+	/* We only want the m and c lines for audio */
+	while ((m = get_sdp_iterate(&miterator, req, "m"))) {
+		if ((media == SDP_AUDIO && ((sscanf(m, "audio %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
+		    (sscanf(m, "audio %d RTP/AVP %n", &x, &len) == 1 && len > 0))) ||
+			(media == SDP_VIDEO && ((sscanf(m, "video %d/%d RTP/AVP %n", &x, &numberofports, &len) == 2 && len > 0) ||
+		    (sscanf(m, "video %d RTP/AVP %n", &x, &len) == 1 && len > 0)))) {
+			/* See if there's a c= line for this media stream.
+			 * XXX There is no guarantee that we'll be grabbing the c= line for this
+			 * particular media stream here. However, this is the same logic used in process_sdp.
+			 */
+			c = get_sdp_iterate(&citerator, req, "c");
+			if (!ast_strlen_zero(c)) {
+				sscanf(c, "IN IP4 %256s", host);
+			}
+			break;
+		}
+	}
+
+	if (ast_strlen_zero(host) || x == 0) {
+		ast_log(LOG_WARNING, "Failed to read an alternate host or port in SDP. Expect %s problems\n", media == SDP_AUDIO ? "audio" : "video");
+		return -1;
+	}
+
+	hp = ast_gethostbyname(host, &audiohp);
+	if (!hp) {
+		ast_log(LOG_WARNING, "Could not look up IP address of alternate hostname. Expect %s problems\n", media == SDP_AUDIO? "audio" : "video");
+		return -1;
+	}
+
+	memcpy(&sin->sin_addr, hp->h_addr, sizeof(sin->sin_addr));
+	sin->sin_port = htons(x);
+	return 0;
+}
+
 /*! \brief Process SIP SDP offer, select formats and activate RTP channels
 	If offer is rejected, we will not change any properties of the call
  	Return 0 on success, a negative value on errors.
@@ -14459,6 +14516,21 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
 		} else {
 			/* We already have a pending invite. Sorry. You are on hold. */
 			p->glareinvite = seqno;
+			if (p->rtp && find_sdp(req)) {
+				struct sockaddr_in sin;
+				if (get_ip_and_port_from_sdp(req, SDP_AUDIO, &sin)) {
+					ast_log(LOG_WARNING, "Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call.\n");
+				} else {
+					ast_rtp_set_alt_peer(p->rtp, &sin);
+				}
+				if (p->vrtp) {
+					if (get_ip_and_port_from_sdp(req, SDP_VIDEO, &sin)) {
+						ast_log(LOG_WARNING, "Failed to set an alternate media source on glared reinvite. Video may not work properly on this call.\n");
+					} else {
+						ast_rtp_set_alt_peer(p->vrtp, &sin);
+					}
+				}
+			}
 			transmit_response_reliable(p, "491 Request Pending", req);
 			if (option_debug)
 				ast_log(LOG_DEBUG, "Got INVITE on call where we already have pending INVITE, deferring that - %s\n", p->callid);