[svn-commits] branch oej/sdpcleanup r27330 - in
/team/oej/sdpcleanup: ./ channels/chan_sip.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue May 16 06:05:01 MST 2006
Author: oej
Date: Tue May 16 08:05:00 2006
New Revision: 27330
URL: http://svn.digium.com/view/asterisk?rev=27330&view=rev
Log:
Update
Modified:
team/oej/sdpcleanup/ (props changed)
team/oej/sdpcleanup/channels/chan_sip.c
Propchange: team/oej/sdpcleanup/
------------------------------------------------------------------------------
automerge = http://edvina.net/training/
Modified: team/oej/sdpcleanup/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/sdpcleanup/channels/chan_sip.c?rev=27330&r1=27329&r2=27330&view=diff
==============================================================================
--- team/oej/sdpcleanup/channels/chan_sip.c (original)
+++ team/oej/sdpcleanup/channels/chan_sip.c Tue May 16 08:05:00 2006
@@ -3875,12 +3875,11 @@
int destiterator = 0;
int iterator;
int sendonly = 0;
- int x;
int numberofports;
int debug=sip_debug_test_pvt(p);
struct ast_channel *bridgepeer = NULL;
- struct ast_rtp newaudiortp, newvideortp;
- int newjointcapability;
+ struct ast_rtp newaudiortp, newvideortp; /* Buffers for codec handling */
+ int newjointcapability; /* Negotiated capability */
int newpeercapability;
int newnoncodeccapability;
int numberofmediastreams = 0;
@@ -3891,6 +3890,8 @@
}
/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
+ memset(&newaudiortp, 0, sizeof(newaudiortp));
+ memset(&newvideortp, 0, sizeof(newvideortp));
ast_rtp_pt_default(&newaudiortp);
ast_rtp_pt_default(&newvideortp);
@@ -3933,6 +3934,7 @@
/* Find media streams in this SDP offer */
while ((m = get_sdp_iterate(&iterator, req, "m"))[0] != '\0') {
+ int x;
int audio = FALSE;
numberofmediastreams++;
@@ -3967,28 +3969,27 @@
return -1;
}
if (debug)
- ast_verbose("Found RTP video format %d\n", codec);
- ast_rtp_set_m_type(&newvideortp, codec);
- }
- } else
- ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
- if (numberofports > 1)
- ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
+ ast_verbose("Found RTP video format %d\n", codec);
+ ast_rtp_set_m_type(&newvideortp, codec);
+ }
+ } else
+ ast_log(LOG_WARNING, "Unsupported SDP media type in offer: %s\n", m);
+ if (numberofports > 1)
+ ast_log(LOG_WARNING, "SDP offered %d ports for media, not supported by Asterisk. Will try anyway...\n", numberofports);
- /* Check for Media-description-level-address for audio */
- if (pedanticsipchecking) {
- c = get_sdp_iterate(&destiterator, req, "c");
- if (!ast_strlen_zero(c)) {
- if (sscanf(c, "IN IP4 %256s", host) != 1) {
- ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c);
- } else {
- /* XXX This could block for a long time, and block the main thread! XXX */
- if (!audio && !(hp = ast_gethostbyname(host, &audiohp)))
+ /* Check for Media-description-level-address for audio */
+ c = get_sdp_iterate(&destiterator, req, "c");
+ if (!ast_strlen_zero(c)) {
+ if (sscanf(c, "IN IP4 %256s", host) != 1) {
+ ast_log(LOG_WARNING, "Invalid secondary host in c= line, '%s'\n", c);
+ } else {
+ /* XXX This could block for a long time, and block the main thread! XXX */
+ if (audio) {
+ if ( !(hp = ast_gethostbyname(host, &audiohp)))
ast_log(LOG_WARNING, "Unable to lookup RTP Audio host in secondary c= line, '%s'\n", c);
- else if (!(vhp = ast_gethostbyname(host, &videohp)))
- ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
- }
+ } else if (!(vhp = ast_gethostbyname(host, &videohp)))
+ ast_log(LOG_WARNING, "Unable to lookup RTP video host in secondary c= line, '%s'\n", c);
}
}
@@ -4041,6 +4042,11 @@
if (debug)
ast_verbose("Got unsupported a:fmtp in SDP offer \n");
continue;
+ } else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
+ /* Video stuff: Not supported */
+ if (debug)
+ ast_verbose("Got unsupported a:framerate in SDP offer \n");
+ continue;
} else if (!strncasecmp(a, "maxprate:", (size_t) 9)) {
/* Video stuff: Not supported */
if (debug)
@@ -4088,42 +4094,38 @@
if (debug) {
/* shame on whoever coded this.... */
- const unsigned slen=512;
- char s1[slen], s2[slen], s3[slen], s4[slen];
+ char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ];
ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n",
- ast_getformatname_multiple(s1, slen, p->capability),
- ast_getformatname_multiple(s2, slen, newpeercapability),
- ast_getformatname_multiple(s3, slen, vpeercapability),
- ast_getformatname_multiple(s4, slen, newjointcapability));
-
- ast_verbose("Non-codec capabilities: us - %s, peer - %s, combined - %s\n",
- ast_rtp_lookup_mime_multiple(s1, slen, noncodeccapability, 0),
- ast_rtp_lookup_mime_multiple(s2, slen, peernoncodeccapability, 0),
- ast_rtp_lookup_mime_multiple(s3, slen, newnoncodeccapability, 0));
+ ast_getformatname_multiple(s1, BUFSIZ, p->capability),
+ ast_getformatname_multiple(s2, BUFSIZ, newpeercapability),
+ ast_getformatname_multiple(s3, BUFSIZ, vpeercapability),
+ ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
+
+ ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
+ ast_rtp_lookup_mime_multiple(s1, BUFSIZ, noncodeccapability, 0),
+ ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0),
+ ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0));
}
if (!newjointcapability) {
- ast_log(LOG_NOTICE, "No compatible codecs!\n");
+ ast_log(LOG_NOTICE, "No compatible codecs, not accepting this offer!\n");
/* Do NOT Change current setting */
return -1;
}
/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
they are acceptable */
- p->jointcapability = newjointcapability;
- p->peercapability = newpeercapability;
- p->noncodeccapability = newnoncodeccapability;
-
+ p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
+ p->peercapability = newpeercapability; /* The other sides capability in latest offer */
+ p->noncodeccapability = newnoncodeccapability; /* DTMF capabilities */
+
{
int i;
/* Copy payload types from source to destination */
for (i=0; i < MAX_RTP_PT; ++i) {
- p->rtp->current_RTP_PT[i].isAstFormat = newaudiortp.current_RTP_PT[i].isAstFormat;
- p->rtp->current_RTP_PT[i].code = newaudiortp.current_RTP_PT[i].code;
- if (p->vrtp) {
- p->vrtp->current_RTP_PT[i].isAstFormat = newvideortp.current_RTP_PT[i].isAstFormat;
- p->vrtp->current_RTP_PT[i].code = newvideortp.current_RTP_PT[i].code;
- }
+ p->rtp->current_RTP_PT[i]= newaudiortp.current_RTP_PT[i];
+ if (p->vrtp)
+ p->vrtp->current_RTP_PT[i]= newvideortp.current_RTP_PT[i];
}
}
@@ -4142,7 +4144,7 @@
}
/* Ok, we're going with this offer */
- if (option_debug) {
+ if (option_debug > 1) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->capability));
}
@@ -4152,12 +4154,12 @@
if (!(p->owner->nativeformats & p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
- const unsigned slen=512;
- char s1[slen], s2[slen];
- if (debug)
- ast_log(LOG_DEBUG, "Oooh, we need to change our formats since our peer supports only %s and not %s\n",
- ast_getformatname_multiple(s1, slen, p->jointcapability),
- ast_getformatname_multiple(s2, slen, p->owner->nativeformats));
+ if (debug) {
+ char s1[BUFSIZ], s2[BUFSIZ];
+ ast_log(LOG_DEBUG, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n",
+ ast_getformatname_multiple(s1, BUFSIZ, p->jointcapability),
+ ast_getformatname_multiple(s2, BUFSIZ, p->owner->nativeformats));
+ }
p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability);
ast_set_read_format(p->owner, p->owner->readformat);
ast_set_write_format(p->owner, p->owner->writeformat);
@@ -4867,7 +4869,8 @@
char o[256];
char c[256];
char t[256];
- char b[256];
+ char b[256] = "";
+ char hold[256] = "";
char m_audio[256];
char m_video[256];
char a_audio[1024];
@@ -4917,7 +4920,7 @@
dest.sin_port = sin.sin_port;
}
- if(capability & AST_FORMAT_VIDEO_MASK) {
+ if((capability & AST_FORMAT_VIDEO_MASK) && ast_test_flag(&p->flags[0], SIP_NOVIDEO))) {
if (p->vrtp) {
needvideo = TRUE;
if (option_debug)
@@ -4927,8 +4930,8 @@
}
- /* Determine video destination */
if (needvideo) {
+ /* Determine video destination */
if (p->vredirip.sin_addr.s_addr) {
vdest.sin_port = p->vredirip.sin_port;
vdest.sin_addr = p->vredirip.sin_addr;
@@ -4936,12 +4939,13 @@
vdest.sin_addr = p->ourip;
vdest.sin_port = vsin.sin_port;
}
- }
- if (debug) {
- ast_verbose("We're at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(sin.sin_port));
- if (needvideo)
+ /* Build max bitrate string */
+ 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));
}
+ if (debug)
+ ast_verbose("Audio is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(sin.sin_port));
/* We break with the "recommendation" and send our IP, in order that our
peer doesn't have to ast_gethostbyname() us */
@@ -4950,10 +4954,7 @@
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 ((p->vrtp) &&
- (!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
- (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
- snprintf(b, sizeof(b), "b=CT:%d\r\n", p->maxcallbitrate);
+ 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));
@@ -5000,12 +5001,12 @@
/* Now send any other common codecs, and non-codec formats: */
for (x = 1;
- x <= ((ast_test_flag(&p->flags[1], SIP_PAGE2_VIDEOSUPPORT) && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO);
+ x <= needvideo ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO;
x <<= 1) {
- if (!(capability & x))
+ if (!(capability & x)) /* Codec not requested */
continue;
- if (alreadysent & x)
+ if (alreadysent & x) /* Already added to SDP */
continue;
if (x <= AST_FORMAT_MAX_AUDIO)
@@ -5039,11 +5040,14 @@
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);
- if ((p->vrtp) &&
- (!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
- (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
- len += strlen(m_video) + strlen(a_video) + strlen(b);
+ if (ast_test_flag(&p->flags[0], SIP_CALL_ONHOLD))
+ sprintf(hold, "a=recvonly");
+ else
+ sprintf(hold, "a=sendrecv");
+
+ len = strlen(v) + strlen(s) + 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);
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
@@ -5051,18 +5055,16 @@
add_line(resp, o);
add_line(resp, s);
add_line(resp, c);
- if ((needvideo) &&
- (!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
- (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
+ if (needvideo) /* only if video response is appropriate */
add_line(resp, b);
add_line(resp, t);
add_line(resp, m_audio);
add_line(resp, a_audio);
- if ((p->vrtp) &&
- (!ast_test_flag(&p->flags[0], SIP_NOVIDEO)) &&
- (capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */
+ add_line(resp, hold);
+ if (needvideo) { /* only if video response is appropriate */
add_line(resp, m_video);
add_line(resp, a_video);
+ add_line(resp, hold);
}
/* Update lastrtprx when we send our SDP */
@@ -10068,14 +10070,14 @@
}
}
-/*! \brief Handle SIP response in dialogue */
+/*! \brief Handle SIP response to INVITE dialogue */
static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, struct sip_request *req, int seqno)
{
int outgoing = ast_test_flag(&p->flags[0], SIP_OUTGOING);
int res = 0;
+ int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
if (option_debug > 3) {
- int reinvite = (p->owner && p->owner->_state == AST_STATE_UP);
if (reinvite)
ast_log(LOG_DEBUG, "SIP response %d to RE-invite on %s call %s\n", resp, outgoing ? "outgoing" : "incoming", p->callid);
else
@@ -10133,8 +10135,10 @@
p->authtries = 0;
if (!strcasecmp(get_header(req, "Content-Type"), "application/sdp")) {
if ((res = process_sdp(p, req)) && !ast_test_flag(req, SIP_PKT_IGNORE))
- /* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
- ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
+ if (!reinvite)
+ /* This 200 OK's SDP is not acceptable, so we need to ack, then hangup */
+ /* For re-invites, we try to recover */
+ ast_set_flag(&p->flags[0], SIP_PENDINGBYE);
}
/* Parse contact header for continued conversation */
@@ -10159,7 +10163,7 @@
}
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
- if (p->owner->_state != AST_STATE_UP) {
+ if (!reinvite)
ast_queue_control(p->owner, AST_CONTROL_ANSWER);
} else { /* RE-invite */
ast_queue_frame(p->owner, &ast_null_frame);
@@ -13851,11 +13855,13 @@
changed = 1;
}
if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
- if (chan->_state != AST_STATE_UP) {
+ if (chan->_state != AST_STATE_UP) { /* We are in early state */
char iabuf[INET_ADDRSTRLEN];
+ if (recordhistory)
+ append_history(p, "ExtInv", "Initial invite sent with remote bridge proposal.");
if (option_debug)
ast_log(LOG_DEBUG, "Early remote bridge setting SIP '%s' - Sending media to %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
- } else if (!p->pendinginvite) {
+ } else if (!p->pendinginvite) { /* We are up, and have no outstanding invite */
if (option_debug > 2) {
char iabuf[INET_ADDRSTRLEN];
ast_log(LOG_DEBUG, "Sending reinvite on SIP '%s' - It's audio soon redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
@@ -13866,6 +13872,7 @@
char iabuf[INET_ADDRSTRLEN];
ast_log(LOG_DEBUG, "Deferring reinvite on SIP '%s' - It's audio will be redirected to IP %s\n", p->callid, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp ? p->redirip.sin_addr : p->ourip));
}
+ /* We have a pending Invite. Send re-invite when we're done with the invite */
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
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