[asterisk-commits] branch oej/sdpcleanup r32303 -
/team/oej/sdpcleanup/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Mon Jun 5 08:26:42 MST 2006
Author: oej
Date: Mon Jun 5 10:26:42 2006
New Revision: 32303
URL: http://svn.digium.com/view/asterisk?rev=32303&view=rev
Log:
Cleanup, remove debug comments
Modified:
team/oej/sdpcleanup/channels/chan_sip.c
Modified: team/oej/sdpcleanup/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/sdpcleanup/channels/chan_sip.c?rev=32303&r1=32302&r2=32303&view=diff
==============================================================================
--- team/oej/sdpcleanup/channels/chan_sip.c (original)
+++ team/oej/sdpcleanup/channels/chan_sip.c Mon Jun 5 10:26:42 2006
@@ -3286,7 +3286,7 @@
/* Set the native formats for audio and merge in video */
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
- if (option_debug) {
+ if (option_debug > 2) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability));
@@ -4076,7 +4076,9 @@
return 0;
}
-/*! \brief Process SIP SDP, select formats and activate RTP channels */
+/*! \brief Process SIP SDP offer, select formats and activate RTP channels
+ If offer is rejected, we will not change any properties of the call
+*/
static int process_sdp(struct sip_pvt *p, struct sip_request *req)
{
const char *m; /* SDP media offer */
@@ -5069,19 +5071,24 @@
{
int len = 0;
int alreadysent = 0;
+
struct sockaddr_in sin;
struct sockaddr_in vsin;
- char v[10];
- char s[256];
+ struct sockaddr_in dest;
+ struct sockaddr_in vdest = { 0, };
+
+ /* SDP fields */
+ char *v = "v=0\r\n"; /* Version */
+ char *subject = "s=session\r\n"; /* Subject of the session */
char o[256];
- char c[256];
- char t[256] = "";
- char b[256] = "";
+ char c[256]; /* Connection data */
+ char *t = "";
+ char b[256] = ""; /* Max bitrate */
char *hold;
- char m_audio[256];
- char m_video[256];
- char a_audio[1024];
- char a_video[1024];
+ char m_audio[256]; /* Media declaration line for audio */
+ char m_video[256]; /* Media declaration line for video */
+ char a_audio[1024]; /* Attributes for audio */
+ char a_video[1024]; /* Attributes for video */
char *m_audio_next = m_audio;
char *m_video_next = m_video;
size_t m_audio_left = sizeof(m_audio);
@@ -5090,13 +5097,14 @@
char *a_video_next = a_video;
size_t a_audio_left = sizeof(a_audio);
size_t a_video_left = sizeof(a_video);
+
char iabuf[INET_ADDRSTRLEN];
int x;
int capability;
- struct sockaddr_in dest;
- struct sockaddr_in vdest = { 0, };
int debug;
int needvideo = FALSE;
+
+ m_video[0] = '\0'; /* Reset the video media string if it's not needed */
debug = sip_debug_test_pvt(p);
@@ -5105,13 +5113,15 @@
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
return -1;
}
- capability = p->jointcapability;
-
+
+ /* Set RTP Session ID and version */
if (!p->sessionid) {
p->sessionid = getpid();
p->sessionversion = p->sessionid;
} else
p->sessionversion++;
+
+ /* Get our addresses */
ast_rtp_get_us(p->rtp, &sin);
if (p->vrtp)
ast_rtp_get_us(p->vrtp, &vsin);
@@ -5126,12 +5136,17 @@
dest.sin_addr = p->ourip;
dest.sin_port = sin.sin_port;
}
+
+ /* Ok, let's start working with codec selection here */
+ capability = p->jointcapability;
+
if (option_debug > 1) {
char codecbuf[BUFSIZ];
ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False");
ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
}
+ /* Check if we need video in this call */
if((capability & AST_FORMAT_VIDEO_MASK) && !ast_test_flag(&p->flags[0], SIP_NOVIDEO)) {
if (p->vrtp) {
needvideo = TRUE;
@@ -5142,20 +5157,27 @@
}
+ /* Ok, we need video. Let's add what we need for video and set codecs.
+ Video is handled differently than audio since we can not transcode. */
if (needvideo) {
+ t = "t=0 0\r\n";
+ ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
+
/* Determine video destination */
if (p->vredirip.sin_addr.s_addr) {
+ vdest.sin_addr = p->vredirip.sin_addr;
vdest.sin_port = p->vredirip.sin_port;
- vdest.sin_addr = p->vredirip.sin_addr;
} else {
vdest.sin_addr = p->ourip;
vdest.sin_port = vsin.sin_port;
}
+
/* Build max bitrate string */
if (p->maxcallbitrate)
snprintf(b, sizeof(b), "b=CT:%d\r\n", p->maxcallbitrate);
if (debug)
ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(vsin.sin_port));
+
/* For video, we can't negotiate video offers. Let's compare the incoming call with what we got. */
if (p->prefcodec) {
int videocapability = (capability & p->prefcodec) & AST_FORMAT_VIDEO_MASK; /* Outbound call */
@@ -5171,7 +5193,8 @@
/* Replace video capabilities with the new videocapability */
capability = (capability & AST_FORMAT_AUDIO_MASK) | videocapability;
- if (option_debug > 2) {
+
+ if (option_debug > 4) {
char codecbuf[BUFSIZ];
if (videocapability)
ast_log(LOG_DEBUG, "** Our video codec selection is: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), videocapability));
@@ -5182,25 +5205,29 @@
if (debug)
ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(sin.sin_port));
+ /* Start building generic SDP headers */
+
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
- snprintf(v, sizeof(v), "v=0\r\n");
snprintf(o, sizeof(o), "o=root %d %d IN IP4 %s\r\n", p->sessionid, p->sessionversion, ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr));
- snprintf(s, sizeof(s), "s=session\r\n");
snprintf(c, sizeof(c), "c=IN IP4 %s\r\n", ast_inet_ntoa(iabuf, sizeof(iabuf), dest.sin_addr));
- if (needvideo)
- snprintf(t, sizeof(t), "t=0 0\r\n");
-
ast_build_string(&m_audio_next, &m_audio_left, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
- ast_build_string(&m_video_next, &m_video_left, "m=video %d RTP/AVP", ntohs(vdest.sin_port));
if (ast_test_flag(&p->flags[0], SIP_CALL_ONHOLD))
hold = "a=recvonly\r\n";
else
hold = "a=sendrecv\r\n";
- /* Prefer the audio codec we were requested to use, first, no matter what */
+ /* Now, start adding audio codecs. These are added in this order:
+ - First what was requested by the calling channel
+ - Then preferences in order from sip.conf device config for this peer/user
+ - Then other codecs in capabilities, including video
+ */
+
+ /* Prefer the audio codec we were requested to use, first, no matter what
+ Note that p->prefcodec can include video codecs, so mask them out
+ */
if (capability & p->prefcodec) {
add_codec_to_sdp(p, p->prefcodec & AST_FORMAT_AUDIO_MASK, 8000,
&m_audio_next, &m_audio_left,
@@ -5209,7 +5236,6 @@
alreadysent |= p->prefcodec & AST_FORMAT_AUDIO_MASK;
}
- ast_log(LOG_DEBUG, "****** ONE ************************\n");
/* Start by sending our preferred audio codecs */
for (x = 0; x < 32; x++) {
@@ -5230,7 +5256,6 @@
debug);
alreadysent |= pref_codec;
}
- ast_log(LOG_DEBUG, "****** TWO ************************\n");
/* Now send any other common audio and video codecs, and non-codec formats: */
for (x = 1; x <= (needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
@@ -5247,17 +5272,12 @@
&m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left,
debug);
- else {
+ else
add_codec_to_sdp(p, x, 90000,
&m_video_next, &m_video_left,
&a_video_next, &a_video_left,
debug);
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Adding video codec to SDP... %d\n", x);
- }
- }
-
- ast_log(LOG_DEBUG, "****** THREE ************************\n");
+ }
/* Now add DTMF RFC2833 telephony-event as a codec */
for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
@@ -5269,8 +5289,9 @@
&a_audio_next, &a_audio_left,
debug);
}
+
if (option_debug > 2)
- ast_log(LOG_DEBUG, "Done with adding codecs to SDP\n");
+ ast_log(LOG_DEBUG, "-- Done with adding codecs to SDP\n");
if(!ast_internal_timing_enabled(p->owner))
ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n");
@@ -5279,13 +5300,10 @@
ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
ast_build_string(&m_audio_next, &m_audio_left, "\r\n");
- ast_build_string(&m_video_next, &m_video_left, "\r\n");
-
- len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_audio) + strlen(a_audio) + strlen(hold);
if (needvideo)
- len += strlen(m_video) + strlen(a_video) + strlen(b) + strlen(hold);
-
- len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m_audio) + strlen(a_audio) + strlen(hold);
+ ast_build_string(&m_video_next, &m_video_left, "\r\n");
+
+ len = strlen(v) + strlen(subject) + strlen(o) + strlen(c) + strlen(t) + strlen(m_audio) + strlen(a_audio) + strlen(hold);
if (needvideo) /* only if video response is appropriate */
len += strlen(m_video) + strlen(a_video) + strlen(b) + strlen(hold);
@@ -5293,7 +5311,7 @@
add_header_contentLength(resp, len);
add_line(resp, v);
add_line(resp, o);
- add_line(resp, s);
+ add_line(resp, subject);
add_line(resp, c);
if (needvideo) /* only if video response is appropriate */
add_line(resp, b);
@@ -5304,7 +5322,7 @@
if (needvideo) { /* only if video response is appropriate */
add_line(resp, m_video);
add_line(resp, a_video);
- add_line(resp, hold);
+ add_line(resp, hold); /* Repeat hold for the video stream */
}
/* Update lastrtprx when we send our SDP */
@@ -5312,10 +5330,8 @@
if (option_debug > 2) {
char buf[BUFSIZ];
- ast_log(LOG_DEBUG, "We're settling with offering these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
- }
- if (option_debug > 2)
- ast_log(LOG_DEBUG, "Done building SDP\n");
+ ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
+ }
return 0;
}
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