[asterisk-commits] mmichelson: branch 1.4 r207423 - /branches/1.4/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Jul 20 14:40:05 CDT 2009
Author: mmichelson
Date: Mon Jul 20 14:39:59 2009
New Revision: 207423
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=207423
Log:
Answer video SDP offers properly when videosupport is not enabled.
Copied from Review board:
In issue 12434, the reporter describes a situation in which audio and video
is offered on the call, but because videosupport is disabled in sip.conf,
Asterisk gives no response at all to the video offer. According to RFC 3264,
all media offers should have a corresponding answer. For offers we do not
intend to actually reply to with meaningful values, we should still reply
with the port for the media stream set to 0.
In this patch, we take note of what types of media have been offered and
save the information on the sip_pvt. The SDP in the response will take into
account whether media was offered. If we are not otherwise going to answer
a media offer, we will insert an appropriate m= line with the port set to 0.
It is important to note that this patch is pretty much a bandage being
applied to a broken bone. The patch *only* helps for situations where video
is offered but videosupport is disabled and when udptl_pt is disabled but
T.38 is offered. Asterisk is not guaranteed to respond to every media offer.
Notable cases are when multiple streams of the same type are offered.
The 2 media stream limit is still present with this patch, too.
In trunk and the 1.6.X branches, things will be a bit different since Asterisk
also supports text in SDPs as well.
(closes issue #12434)
Reported by: mnnojd
Review: https://reviewboard.asterisk.org/r/311
Review: https://reviewboard.asterisk.org/r/313
Modified:
branches/1.4/channels/chan_sip.c
Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=207423&r1=207422&r2=207423
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Mon Jul 20 14:39:59 2009
@@ -911,6 +911,10 @@
enum referstatus status; /*!< REFER status */
};
+struct offered_media {
+ int offered;
+ char text[128];
+};
/*! \brief sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe */
static struct sip_pvt {
ast_mutex_t lock; /*!< Dialog private lock */
@@ -1034,6 +1038,21 @@
struct sip_invite_param *options; /*!< Options for INVITE */
int autoframing;
int hangupcause; /*!< Storage of hangupcause copied from our owner before we disconnect from the AST channel (only used at hangup) */
+ /*! When receiving an SDP offer, it is important to take note of what media types were offered.
+ * By doing this, even if we don't want to answer a particular media stream with something meaningful, we can
+ * still put an m= line in our answer with the port set to 0.
+ *
+ * The reason for the length being 3 is that in this branch of Asterisk, the only media types supported are
+ * image, audio, and video. Therefore we need to keep track of which types of media were offered.
+ *
+ * Note that if we wanted to be 100% correct, we would keep a list of all media streams offered. That way we could respond
+ * even to unknown media types, and we could respond to multiple streams of the same type. Such large-scale changes
+ * are not a good idea for released branches, though, so we're compromising by just making sure that for the common cases:
+ * audio and video, and audio and T.38, we give the appropriate response to both media streams.
+ *
+ * The large-scale changes would be a good idea for implementing during an SDP rewrite.
+ */
+ struct offered_media offered_media[3];
} *iflist = NULL;
/*! Max entires in the history list for a sip_pvt */
@@ -5148,6 +5167,7 @@
enum media_type {
SDP_AUDIO,
SDP_VIDEO,
+ SDP_IMAGE,
};
static int get_ip_and_port_from_sdp(struct sip_request *req, const enum media_type media, struct sockaddr_in *sin)
@@ -5274,6 +5294,7 @@
/* Update our last rtprx when we receive an SDP, too */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
+ memset(p->offered_media, 0, sizeof(p->offered_media));
/* Try to find first media stream */
m = get_sdp(req, "m");
@@ -5312,11 +5333,14 @@
if ((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)) {
audio = TRUE;
+ p->offered_media[SDP_AUDIO].offered = TRUE;
numberofmediastreams++;
/* Found audio stream in this media definition */
portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
- for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
+ codecs = m + len;
+ ast_copy_string(p->offered_media[SDP_AUDIO].text, codecs, sizeof(p->offered_media[SDP_AUDIO].text));
+ for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
return -1;
@@ -5328,10 +5352,13 @@
} else if ((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)) {
/* If it is not audio - is it video ? */
- ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
+ ast_clear_flag(&p->flags[0], SIP_NOVIDEO);
+ p->offered_media[SDP_VIDEO].offered = TRUE;
numberofmediastreams++;
vportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
+ codecs = m + len;
+ ast_copy_string(p->offered_media[SDP_VIDEO].text, codecs, sizeof(p->offered_media[SDP_VIDEO].text));
for (codecs = m + len; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@@ -5345,6 +5372,7 @@
(sscanf(m, "image %d UDPTL t38%n", &x, &len) == 1 && len >= 0) )) {
if (debug)
ast_verbose("Got T.38 offer in SDP in dialog %s\n", p->callid);
+ p->offered_media[SDP_IMAGE].offered = TRUE;
udptlportno = x;
numberofmediastreams++;
@@ -6670,6 +6698,7 @@
size_t a_audio_left = sizeof(a_audio);
size_t a_video_left = sizeof(a_video);
size_t a_modem_left = sizeof(a_modem);
+ char dummy_answer[256];
int x;
int capability = 0;
@@ -6756,7 +6785,7 @@
vdest.sin_addr = p->ourip;
vdest.sin_port = vsin.sin_port;
}
- ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
+ ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vsin.sin_port));
/* Build max bitrate string */
if (p->maxcallbitrate)
@@ -6926,15 +6955,23 @@
add_line(resp, m_audio);
add_line(resp, a_audio);
add_line(resp, hold);
+ } else if (p->offered_media[SDP_AUDIO].offered) {
+ snprintf(dummy_answer, sizeof(dummy_answer), "m=audio 0 RTP/AVP %s\r\n", p->offered_media[SDP_AUDIO].text);
+ add_line(resp, dummy_answer);
}
if (needvideo) { /* only if video response is appropriate */
add_line(resp, m_video);
add_line(resp, a_video);
add_line(resp, hold); /* Repeat hold for the video stream */
+ } else if (p->offered_media[SDP_VIDEO].offered) {
+ snprintf(dummy_answer, sizeof(dummy_answer), "m=video 0 RTP/AVP %s\r\n", p->offered_media[SDP_VIDEO].text);
+ add_line(resp, dummy_answer);
}
if (add_t38) {
add_line(resp, m_modem);
add_line(resp, a_modem);
+ } else if (p->offered_media[SDP_IMAGE].offered) {
+ add_line(resp, "m=image 0 udptl t38\r\n");
}
/* Update lastrtprx when we send our SDP */
@@ -7076,6 +7113,7 @@
add_header(&req, "X-asterisk-Info", "SIP re-invite (External RTP bridge)");
if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))
append_history(p, "ReInv", "Re-invite sent");
+ memset(p->offered_media, 0, sizeof(p->offered_media));
add_sdp(&req, p, 1, 0);
/* Use this as the basis */
initialize_initreq(p, &req);
@@ -7098,6 +7136,7 @@
add_header(&req, "Supported", SUPPORTED_EXTENSIONS);
if (sipdebug)
add_header(&req, "X-asterisk-info", "SIP re-invite (T38 switchover)");
+ memset(p->offered_media, 0, sizeof(p->offered_media));
add_sdp(&req, p, 0, 1);
/* Use this as the basis */
@@ -7430,6 +7469,7 @@
ast_channel_unlock(chan);
}
if (sdp) {
+ memset(p->offered_media, 0, sizeof(p->offered_media));
if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
ast_udptl_offered_from_local(p->udptl, 1);
if (option_debug)
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