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authorOlle Johansson <oej@edvina.net>2009-09-07 18:00:48 +0000
committerOlle Johansson <oej@edvina.net>2009-09-07 18:00:48 +0000
commitc55469da80cfe41165202e32df8e4d2ce2579b46 (patch)
tree84563cf6bb875da50849e36c4a681fe29c6637ce /channels
parent75d8960740feb23d0752170e5bcd01d6786c2791 (diff)
Clean up the "offered_media" code
- Add variable for number of known media streams instead of hardcoding in definition of sip_pvt - Rename "text" to "codecs" - beacuse it's what it is - Add documentation for future developers so that we make sure that we define new sdp media types for SRTP-variants git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@216883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c37
1 files changed, 23 insertions, 14 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 3c1e119bf..34140cab0 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -711,6 +711,19 @@ enum subscriptiontype {
MWI_NOTIFICATION
};
+/*! \brief The number of media types in enum \ref media_type below. */
+#define OFFERED_MEDIA_COUNT 4
+
+/*! \brief Media types generate different "dummy answers" for not accepting the offer of
+ a media stream. We need to add definitions for each RTP profile. Secure RTP is not
+ the same as normal RTP and will require a new definition */
+enum media_type {
+ SDP_AUDIO, /*!< RTP/AVP Audio */
+ SDP_VIDEO, /*!< RTP/AVP Video */
+ SDP_IMAGE, /*!< Image udptl, not TCP or RTP */
+ SDP_TEXT, /*!< RTP/AVP Realtime Text */
+};
+
/*! \brief Subscription types that we support. We support
- dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
- SIMPLE presence used for device status
@@ -1653,9 +1666,11 @@ struct sip_st_cfg {
int st_max_se; /*!< Highest threshold for session refresh interval */
};
+/*! \brief Structure for remembering offered media in an INVITE, to make sure we reply
+ to all media streams. In theory. In practise, we try our best. */
struct offered_media {
int offered;
- char text[128];
+ char codecs[128];
};
/*! \brief Structure used for each SIP dialog, ie. a call, a registration, a subscribe.
@@ -1840,7 +1855,7 @@ struct sip_pvt {
*
* The large-scale changes would be a good idea for implementing during an SDP rewrite.
*/
- struct offered_media offered_media[4];
+ struct offered_media offered_media[OFFERED_MEDIA_COUNT];
};
@@ -8047,12 +8062,6 @@ static int find_sdp(struct sip_request *req)
return FALSE;
}
-enum media_type {
- SDP_AUDIO,
- SDP_VIDEO,
- SDP_IMAGE,
- SDP_TEXT,
-};
static int get_ip_and_port_from_sdp(struct sip_request *req, const enum media_type media, struct sockaddr_in *sin)
{
@@ -8309,7 +8318,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
portno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
codecs = m + len;
- ast_copy_string(p->offered_media[SDP_AUDIO].text, codecs, sizeof(p->offered_media[SDP_AUDIO].text));
+ ast_copy_string(p->offered_media[SDP_AUDIO].codecs, codecs, sizeof(p->offered_media[SDP_AUDIO].codecs));
for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%30d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@@ -8329,7 +8338,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
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));
+ ast_copy_string(p->offered_media[SDP_VIDEO].codecs, codecs, sizeof(p->offered_media[SDP_VIDEO].codecs));
for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%30d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@@ -8348,7 +8357,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
tportno = x;
/* Scan through the RTP payload types specified in a "m=" line: */
codecs = m + len;
- ast_copy_string(p->offered_media[SDP_TEXT].text, codecs, sizeof(p->offered_media[SDP_TEXT].text));
+ ast_copy_string(p->offered_media[SDP_TEXT].codecs, codecs, sizeof(p->offered_media[SDP_TEXT].codecs));
for (; !ast_strlen_zero(codecs); codecs = ast_skip_blanks(codecs + len)) {
if (sscanf(codecs, "%30d%n", &codec, &len) != 1) {
ast_log(LOG_WARNING, "Error in codec string '%s'\n", codecs);
@@ -10433,7 +10442,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
add_line(resp, a_audio->str);
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);
+ snprintf(dummy_answer, sizeof(dummy_answer), "m=audio 0 RTP/AVP %s\r\n", p->offered_media[SDP_AUDIO].codecs);
add_line(resp, dummy_answer);
}
if (needvideo) { /* only if video response is appropriate */
@@ -10441,7 +10450,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
add_line(resp, a_video->str);
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);
+ snprintf(dummy_answer, sizeof(dummy_answer), "m=video 0 RTP/AVP %s\r\n", p->offered_media[SDP_VIDEO].codecs);
add_line(resp, dummy_answer);
}
if (needtext) { /* only if text response is appropriate */
@@ -10449,7 +10458,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
add_line(resp, a_text->str);
add_line(resp, hold); /* Repeat hold for the text stream */
} else if (p->offered_media[SDP_TEXT].offered) {
- snprintf(dummy_answer, sizeof(dummy_answer), "m=text 0 RTP/AVP %s\r\n", p->offered_media[SDP_TEXT].text);
+ snprintf(dummy_answer, sizeof(dummy_answer), "m=text 0 RTP/AVP %s\r\n", p->offered_media[SDP_TEXT].codecs);
add_line(resp, dummy_answer);
}
if (add_t38) {