[Asterisk-cvs] asterisk/channels chan_sip.c,1.871,1.872
kpfleming
kpfleming
Thu Sep 29 13:44:28 CDT 2005
Update of /usr/cvsroot/asterisk/channels
In directory mongoose.digium.com:/tmp/cvs-serv14138/channels
Modified Files:
chan_sip.c
Log Message:
ensure that peer codec preferences take precedence over global preferences
clean and simplify SDP-building code
Index: chan_sip.c
===================================================================
RCS file: /usr/cvsroot/asterisk/channels/chan_sip.c,v
retrieving revision 1.871
retrieving revision 1.872
diff -u -d -r1.871 -r1.872
--- chan_sip.c 29 Sep 2005 03:12:46 -0000 1.871
+++ chan_sip.c 29 Sep 2005 17:41:00 -0000 1.872
@@ -1811,6 +1811,7 @@
SIP_PROMISCREDIR | SIP_USEREQPHONE | SIP_DTMF | SIP_NAT | SIP_REINVITE |
SIP_INSECURE_PORT | SIP_INSECURE_INVITE);
r->capability = peer->capability;
+ r->prefs = peer->prefs;
if (r->rtp) {
ast_log(LOG_DEBUG, "Setting NAT on RTP to %d\n", (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE));
ast_rtp_setnat(r->rtp, (ast_test_flag(r, SIP_NAT) & SIP_NAT_ROUTE));
@@ -2959,11 +2960,9 @@
{
struct sip_pvt *p;
- p = malloc(sizeof(struct sip_pvt));
- if (!p)
+ if (!(p = calloc(1, sizeof(*p))))
return NULL;
- /* Keep track of stuff */
- memset(p, 0, sizeof(struct sip_pvt));
+
ast_mutex_init(&p->lock);
p->method = intended_method;
@@ -4171,14 +4170,49 @@
return 0;
}
+static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+ char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
+ int debug)
+{
+ int rtp_code;
+
+ if (debug)
+ ast_verbose("Adding codec 0x%x (%s) to SDP\n", codec, ast_getformatname(codec));
+ if ((rtp_code = ast_rtp_lookup_code(p->rtp, 1, codec)) == -1)
+ return;
+
+ ast_build_string(m_buf, m_size, " %d", rtp_code);
+ ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
+ ast_rtp_lookup_mime_subtype(1, codec),
+ sample_rate);
+}
+
+static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
+ char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
+ int debug)
+{
+ int rtp_code;
+
+ if (debug)
+ ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype(0, format));
+ if ((rtp_code = ast_rtp_lookup_code(p->rtp, 0, format)) == -1)
+ return;
+
+ ast_build_string(m_buf, m_size, " %d", rtp_code);
+ ast_build_string(a_buf, a_size, "a=rtpmap:%d %s/%d\r\n", rtp_code,
+ ast_rtp_lookup_mime_subtype(0, format),
+ sample_rate);
+ if (format == AST_RTP_DTMF)
+ /* Indicate we support DTMF and FLASH... */
+ ast_build_string(a_buf, a_size, "a=fmtp:%d 0-16\r\n", rtp_code);
+}
+
/*--- add_sdp: Add Session Description Protocol message ---*/
static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
{
int len = 0;
- int codec = 0;
- int pref_codec = 0;
+ int pref_codec;
int alreadysent = 0;
- char costr[80];
struct sockaddr_in sin;
struct sockaddr_in vsin;
char v[256];
@@ -4186,21 +4220,27 @@
char o[256];
char c[256];
char t[256];
- char m[256];
- char m2[256];
- char a[1024] = "";
- char a2[1024] = "";
+ char m_audio[256];
+ char m_video[256];
+ char a_audio[1024];
+ char a_video[1024];
+ char *m_audio_next = m_audio;
+ char *m_video_next = m_video;
+ size_t m_audio_left = sizeof(m_audio);
+ size_t m_video_left = sizeof(m_video);
+ char *a_audio_next = a_audio;
+ 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 = 0;
- int capability = 0 ;
+ int x;
+ int capability;
struct sockaddr_in dest;
struct sockaddr_in vdest = { 0, };
- int debug=0;
+ int debug;
debug = sip_debug_test_pvt(p);
- /* XXX We break with the "recommendation" and send our IP, in order that our
- peer doesn't have to ast_gethostbyname() us XXX */
len = 0;
if (!p->rtp) {
ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
@@ -4242,105 +4282,100 @@
if (p->vrtp)
ast_verbose("Video is at %s port %d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip), ntohs(vsin.sin_port));
}
+
+ /* 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));
snprintf(t, sizeof(t), "t=0 0\r\n");
- snprintf(m, sizeof(m), "m=audio %d RTP/AVP", ntohs(dest.sin_port));
- snprintf(m2, sizeof(m2), "m=video %d RTP/AVP", ntohs(vdest.sin_port));
+
+ 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));
+
/* Prefer the codec we were requested to use, first, no matter what */
if (capability & p->prefcodec) {
- if (debug)
- ast_verbose("Answering/Requesting with root capability 0x%x (%s)\n", p->prefcodec, ast_getformatname(p->prefcodec));
- codec = ast_rtp_lookup_code(p->rtp, 1, p->prefcodec);
- if (codec > -1) {
- snprintf(costr, sizeof(costr), " %d", codec);
- if (p->prefcodec <= AST_FORMAT_MAX_AUDIO) {
- strncat(m, costr, sizeof(m) - strlen(m) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, p->prefcodec));
- ast_copy_string(a, costr, sizeof(a));
- } else {
- strncat(m2, costr, sizeof(m2) - strlen(m2) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, p->prefcodec));
- ast_copy_string(a2, costr, sizeof(a2));
- }
- }
+ if (p->prefcodec <= AST_FORMAT_MAX_AUDIO)
+ add_codec_to_sdp(p, p->prefcodec, 8000,
+ &m_audio_next, &m_audio_left,
+ &a_audio_next, &a_audio_left,
+ debug);
+ else
+ add_codec_to_sdp(p, p->prefcodec, 90000,
+ &m_video_next, &m_video_left,
+ &a_video_next, &a_video_left,
+ debug);
alreadysent |= p->prefcodec;
}
+
/* Start by sending our preferred codecs */
- for (x = 0 ; x < 32 ; x++) {
- if (!(pref_codec = ast_codec_pref_index(&p->prefs,x)))
+ for (x = 0; x < 32; x++) {
+ if (!(pref_codec = ast_codec_pref_index(&p->prefs, x)))
break;
- if ((capability & pref_codec) && !(alreadysent & pref_codec)) {
- if (debug)
- ast_verbose("Answering with preferred capability 0x%x (%s)\n", pref_codec, ast_getformatname(pref_codec));
- codec = ast_rtp_lookup_code(p->rtp, 1, pref_codec);
- if (codec > -1) {
- snprintf(costr, sizeof(costr), " %d", codec);
- if (pref_codec <= AST_FORMAT_MAX_AUDIO) {
- strncat(m, costr, sizeof(m) - strlen(m) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, pref_codec));
- strncat(a, costr, sizeof(a) - strlen(a) - 1);
- } else {
- strncat(m2, costr, sizeof(m2) - strlen(m2) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, pref_codec));
- strncat(a2, costr, sizeof(a2) - strlen(a) - 1);
- }
- }
- }
+
+ if (!(capability & pref_codec))
+ continue;
+
+ if (alreadysent & pref_codec)
+ continue;
+
+ if (pref_codec <= AST_FORMAT_MAX_AUDIO)
+ add_codec_to_sdp(p, pref_codec, 8000,
+ &m_audio_next, &m_audio_left,
+ &a_audio_next, &a_audio_left,
+ debug);
+ else
+ add_codec_to_sdp(p, pref_codec, 90000,
+ &m_video_next, &m_video_left,
+ &a_video_next, &a_video_left,
+ debug);
alreadysent |= pref_codec;
}
/* Now send any other common codecs, and non-codec formats: */
for (x = 1; x <= ((videosupport && p->vrtp) ? AST_FORMAT_MAX_VIDEO : AST_FORMAT_MAX_AUDIO); x <<= 1) {
- if ((capability & x) && !(alreadysent & x)) {
- if (debug)
- ast_verbose("Answering with capability 0x%x (%s)\n", x, ast_getformatname(x));
- codec = ast_rtp_lookup_code(p->rtp, 1, x);
- if (codec > -1) {
- snprintf(costr, sizeof(costr), " %d", codec);
- if (x <= AST_FORMAT_MAX_AUDIO) {
- strncat(m, costr, sizeof(m) - strlen(m) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x));
- strncat(a, costr, sizeof(a) - strlen(a) - 1);
- } else {
- strncat(m2, costr, sizeof(m2) - strlen(m2) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/90000\r\n", codec, ast_rtp_lookup_mime_subtype(1, x));
- strncat(a2, costr, sizeof(a2) - strlen(a2) - 1);
- }
- }
- }
+ if (!(capability & x))
+ continue;
+
+ if (alreadysent & x)
+ continue;
+
+ if (x <= AST_FORMAT_MAX_AUDIO)
+ add_codec_to_sdp(p, x, 8000,
+ &m_audio_next, &m_audio_left,
+ &a_audio_next, &a_audio_left,
+ debug);
+ else
+ add_codec_to_sdp(p, x, 90000,
+ &m_video_next, &m_video_left,
+ &a_video_next, &a_video_left,
+ debug);
}
+
for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
- if (p->noncodeccapability & x) {
- if (debug)
- ast_verbose("Answering with non-codec capability 0x%x (%s)\n", x, ast_rtp_lookup_mime_subtype(0, x));
- codec = ast_rtp_lookup_code(p->rtp, 0, x);
- if (codec > -1) {
- snprintf(costr, sizeof(costr), " %d", codec);
- strncat(m, costr, sizeof(m) - strlen(m) - 1);
- snprintf(costr, sizeof(costr), "a=rtpmap:%d %s/8000\r\n", codec, ast_rtp_lookup_mime_subtype(0, x));
- strncat(a, costr, sizeof(a) - strlen(a) - 1);
- if (x == AST_RTP_DTMF) {
- /* Indicate we support DTMF and FLASH... */
- snprintf(costr, sizeof costr, "a=fmtp:%d 0-16\r\n",
- codec);
- strncat(a, costr, sizeof(a) - strlen(a) - 1);
- }
- }
- }
+ if (!(p->noncodeccapability & x))
+ continue;
+
+ add_noncodec_to_sdp(p, x, 8000,
+ &m_audio_next, &m_audio_left,
+ &a_audio_next, &a_audio_left,
+ debug);
}
- strncat(a, "a=silenceSupp:off - - - -\r\n", sizeof(a) - strlen(a) - 1);
- if (strlen(m) < sizeof(m) - 2)
- strncat(m, "\r\n", sizeof(m) - strlen(m) - 1);
- if (strlen(m2) < sizeof(m2) - 2)
- strncat(m2, "\r\n", sizeof(m2) - strlen(m2) - 1);
- if ((sizeof(m) <= strlen(m) - 2) || (sizeof(m2) <= strlen(m2) - 2) || (sizeof(a) == strlen(a)) || (sizeof(a2) == strlen(a2)))
+
+ ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - -\r\n");
+
+ if ((m_audio_left < 2) || (m_video_left < 2) || (a_audio_left == 0) || (a_video_left == 0))
ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
- len = strlen(v) + strlen(s) + strlen(o) + strlen(c) + strlen(t) + strlen(m) + strlen(a);
+
+ 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, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) /* only if video response is appropriate */
- len += strlen(m2) + strlen(a2);
+ len += strlen(m_video) + strlen(a_video);
+
add_header(resp, "Content-Type", "application/sdp");
add_header_contentLength(resp, len);
add_line(resp, v);
@@ -4348,15 +4383,17 @@
add_line(resp, s);
add_line(resp, c);
add_line(resp, t);
- add_line(resp, m);
- add_line(resp, a);
+ add_line(resp, m_audio);
+ add_line(resp, a_audio);
if ((p->vrtp) && (!ast_test_flag(p, SIP_NOVIDEO)) && (capability & VIDEO_CODEC_MASK)) { /* only if video response is appropriate */
- add_line(resp, m2);
- add_line(resp, a2);
+ add_line(resp, m_video);
+ add_line(resp, a_video);
}
+
/* Update lastrtprx when we send our SDP */
time(&p->lastrtprx);
time(&p->lastrtptx);
+
return 0;
}
@@ -7006,6 +7043,7 @@
p->callgroup = peer->callgroup;
p->pickupgroup = peer->pickupgroup;
p->capability = peer->capability;
+ p->prefs = peer->prefs;
p->jointcapability = peer->capability;
if (p->peercapability)
p->jointcapability &= p->peercapability;
@@ -11251,9 +11289,6 @@
}
}
- /* Assign a default capability */
- p->capability = global_capability;
-
if (create_addr(p, host)) {
*cause = AST_CAUSE_UNREGISTERED;
sip_destroy(p);
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