[asterisk-commits] oej: trunk r216883 - /trunk/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Sep 7 13:00:51 CDT 2009
Author: oej
Date: Mon Sep 7 13:00:48 2009
New Revision: 216883
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=216883
Log:
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
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.asterisk.org/svn-view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=216883&r1=216882&r2=216883
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Sep 7 13:00:48 2009
@@ -711,6 +711,19 @@
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 @@
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 @@
*
* 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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) {
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