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Diffstat (limited to 'channels/chan_sip.c')
-rw-r--r--channels/chan_sip.c166
1 files changed, 108 insertions, 58 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index f0a7fb9e3..a81b56289 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -2286,10 +2286,10 @@ static const char* get_sdp_iterate(int* start, struct sip_request *req, const ch
static const char *get_sdp(struct sip_request *req, const char *name);
static int find_sdp(struct sip_request *req);
static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action);
-static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size);
-static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
+static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp);
@@ -7671,22 +7671,17 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
iterator = req->sdp_start;
while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
char mimeSubtype[128];
+ char fmtp_string[64];
+ unsigned int sample_rate;
+
if (option_debug > 1) {
int breakout = FALSE;
-
+
/* If we're debugging, check for unsupported sdp options */
if (!strncasecmp(a, "rtcp:", (size_t) 5)) {
if (debug)
ast_verbose("Got unsupported a:rtcp in SDP offer \n");
breakout = TRUE;
- } else if (!strncasecmp(a, "fmtp:", (size_t) 5)) {
- /* Format parameters: Not supported */
- /* Note: This is used for codec parameters, like bitrate for
- G722 and video formats for H263 and H264
- See RFC2327 for an example */
- if (debug)
- ast_verbose("Got unsupported a:fmtp in SDP offer \n");
- breakout = TRUE;
} else if (!strncasecmp(a, "framerate:", (size_t) 10)) {
/* Video stuff: Not supported */
if (debug)
@@ -7706,21 +7701,29 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
if (breakout) /* We have a match, skip to next header */
continue;
}
+
if (!strcasecmp(a, "sendonly")) {
if (sendonly == -1)
sendonly = 1;
continue;
- } else if (!strcasecmp(a, "inactive")) {
+ }
+
+ if (!strcasecmp(a, "inactive")) {
if (sendonly == -1)
sendonly = 2;
continue;
- } else if (!strcasecmp(a, "sendrecv")) {
+ }
+
+ if (!strcasecmp(a, "sendrecv")) {
if (sendonly == -1)
sendonly = 0;
continue;
- } else if (strlen(a) > 5 && !strncasecmp(a, "ptime", 5)) {
+ }
+
+ if (!strncasecmp(a, "ptime", 5)) {
char *tmp = strrchr(a, ':');
long int framing = 0;
+
if (tmp) {
tmp++;
framing = strtol(tmp, NULL, 10);
@@ -7744,8 +7747,9 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
ast_rtp_codec_setpref(p->rtp, pref);
}
continue;
-
- } else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
+ }
+
+ if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
/* count numbers of generations in fmtp */
red_cp = &red_fmtp[strlen(red_fmtp)];
strncpy(red_fmtp, a, 100);
@@ -7757,15 +7761,59 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
red_cp = strtok(NULL, "/");
}
red_cp = red_fmtp;
+ continue;
+ }
+
+ if (sscanf(a, "fmtp: %u %63s", &codec, fmtp_string) == 2) {
+ struct rtpPayloadType payload;
+ unsigned int handled = 0;
+
+ payload = ast_rtp_lookup_pt(newaudiortp, codec);
+ if (!payload.code) {
+ /* it wasn't found, try the video rtp */
+ payload = ast_rtp_lookup_pt(newvideortp, codec);
+ }
+ if (payload.code && payload.isAstFormat) {
+ unsigned int bit_rate;
+
+ switch (payload.code) {
+ case AST_FORMAT_SIREN7:
+ if (sscanf(fmtp_string, "bitrate=%u", &bit_rate) == 1) {
+ if (bit_rate != 32000) {
+ ast_log(LOG_WARNING, "Got Siren7 offer at %d bps, but only 32000 bps supported; ignoring.\n", bit_rate);
+ ast_rtp_unset_m_type(newaudiortp, codec);
+ } else {
+ handled = 1;
+ }
+ }
+ break;
+ case AST_FORMAT_SIREN14:
+ if (sscanf(fmtp_string, "bitrate=%u", &bit_rate) == 1) {
+ if (bit_rate != 48000) {
+ ast_log(LOG_WARNING, "Got Siren14 offer at %d bps, but only 48000 bps supported; ignoring.\n", bit_rate);
+ ast_rtp_unset_m_type(newaudiortp, codec);
+ } else {
+ handled = 1;
+ }
+ }
+ break;
+ }
+ }
+
+ if (!handled) {
+ ast_debug(1, "Got unsupported a:%s in SDP offer\n", a);
+ }
+ continue;
+ }
- } else if (sscanf(a, "rtpmap: %u %127[^/]/", &codec, mimeSubtype) == 2) {
+ if (sscanf(a, "rtpmap: %u %127[^/]/%u", &codec, mimeSubtype, &sample_rate) == 3) {
/* We have a rtpmap to handle */
if (last_rtpmap_codec < SDP_MAX_RTPMAP_CODECS) {
- /* Note: should really look at the 'freq' and '#chans' params too */
+ /* Note: should really look at the '#chans' params too */
/* Note: This should all be done in the context of the m= above */
if (!strncasecmp(mimeSubtype, "H26", 3) || !strncasecmp(mimeSubtype, "MP4", 3)) { /* Video */
- if(ast_rtp_set_rtpmap_type(newvideortp, codec, "video", mimeSubtype, 0) != -1) {
+ if (ast_rtp_set_rtpmap_type_rate(newvideortp, codec, "video", mimeSubtype, 0, sample_rate) != -1) {
if (debug)
ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
@@ -7787,11 +7835,12 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
sprintf(red_fmtp, "fmtp:%d ", red_pt);
if (debug)
- ast_verbose("Red submimetype has payload type: %d\n", red_pt);
+ ast_verbose("RED submimetype has payload type: %d\n", red_pt);
}
} else { /* Must be audio?? */
- if(ast_rtp_set_rtpmap_type(newaudiortp, codec, "audio", mimeSubtype,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0) != -1) {
+ if (ast_rtp_set_rtpmap_type_rate(newaudiortp, codec, "audio", mimeSubtype,
+ ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0,
+ sample_rate) != -1) {
if (debug)
ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
found_rtpmap_codecs[last_rtpmap_codec] = codec;
@@ -8895,7 +8944,7 @@ static int add_vidupdate(struct sip_request *req)
}
/*! \brief Add codec offer to SDP offer/answer body in INVITE or 200 OK */
-static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+static void add_codec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
@@ -8915,18 +8964,31 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec,
- ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
- sample_rate);
- if (codec == AST_FORMAT_G729A) {
+ ast_rtp_lookup_mime_subtype(1, codec,
+ ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
+ ast_rtp_lookup_sample_rate(1, codec));
+
+ switch (codec) {
+ case AST_FORMAT_G729A:
/* Indicate that we don't support VAD (G.729 annex B) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexb=no\r\n", rtp_code);
- } else if (codec == AST_FORMAT_G723_1) {
+ break;
+ case AST_FORMAT_G723_1:
/* Indicate that we don't support VAD (G.723.1 annex A) */
ast_str_append(a_buf, 0, "a=fmtp:%d annexa=no\r\n", rtp_code);
- } else if (codec == AST_FORMAT_ILBC) {
+ break;
+ case AST_FORMAT_ILBC:
/* Add information about us using only 20/30 ms packetization */
ast_str_append(a_buf, 0, "a=fmtp:%d mode=%d\r\n", rtp_code, fmt.cur_ms);
+ break;
+ case AST_FORMAT_SIREN7:
+ /* Indicate that we only expect 32Kbps */
+ ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=32000\r\n", rtp_code);
+ break;
+ case AST_FORMAT_SIREN14:
+ /* Indicate that we only expect 48Kbps */
+ ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=48000\r\n", rtp_code);
+ break;
}
if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
@@ -8939,7 +9001,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate
/*! \brief Add video codec offer to SDP offer/answer body in INVITE or 200 OK */
/* This is different to the audio one now so we can add more caps later */
-static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
@@ -8956,12 +9018,13 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rat
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
+ ast_rtp_lookup_mime_subtype(1, codec, 0),
+ ast_rtp_lookup_sample_rate(1, codec));
/* Add fmtp code here */
}
/*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */
-static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rate,
+static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug, int *min_packet_size)
{
@@ -8978,11 +9041,12 @@ static void add_tcodec_to_sdp(const struct sip_pvt *p, int codec, int sample_rat
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(1, codec, 0), sample_rate);
+ ast_rtp_lookup_mime_subtype(1, codec, 0),
+ ast_rtp_lookup_sample_rate(1, codec));
/* Add fmtp code here */
if (codec == AST_FORMAT_T140RED) {
- ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
+ ast_str_append(a_buf, 0, "a=fmtp:%d %d/%d/%d\r\n", rtp_code,
ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140),
ast_rtp_lookup_code(p->trtp, 1, AST_FORMAT_T140));
@@ -9107,7 +9171,7 @@ static int add_t38_sdp(struct sip_request *resp, struct sip_pvt *p)
/*! \brief Add RFC 2833 DTMF offer to SDP */
-static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
+static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
struct ast_str **m_buf, struct ast_str **a_buf,
int debug)
{
@@ -9120,8 +9184,8 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
- ast_rtp_lookup_mime_subtype(0, format, 0),
- sample_rate);
+ ast_rtp_lookup_mime_subtype(0, format, 0),
+ ast_rtp_lookup_sample_rate(0, format));
if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */
ast_str_append(a_buf, 0, "a=fmtp:%d 0-16\r\n", rtp_code);
}
@@ -9162,13 +9226,6 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo,
}
-/*!
- * \note G.722 actually is supposed to specified as 8 kHz, even though it is
- * really 16 kHz. Update this macro for other formats as they are added in
- * the future.
- */
-#define SDP_SAMPLE_RATE(x) 8000
-
/*! \brief Add Session Description Protocol message
If oldsdp is TRUE, then the SDP version number is not incremented. This mechanism
@@ -9340,9 +9397,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
if (capability & p->prefcodec) {
int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
- add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec),
- &m_audio, &a_audio,
- debug, &min_audio_packet_size);
+ add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
alreadysent |= codec;
}
@@ -9359,9 +9414,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
if (alreadysent & codec)
continue;
- add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec),
- &m_audio, &a_audio,
- debug, &min_audio_packet_size);
+ add_codec_to_sdp(p, codec, &m_audio, &a_audio, debug, &min_audio_packet_size);
alreadysent |= codec;
}
@@ -9374,14 +9427,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
continue;
if (x & AST_FORMAT_AUDIO_MASK)
- add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
- &m_audio, &a_audio, debug, &min_audio_packet_size);
- else if (x & AST_FORMAT_VIDEO_MASK)
- add_vcodec_to_sdp(p, x, 90000,
- &m_video, &a_video, debug, &min_video_packet_size);
+ add_codec_to_sdp(p, x, &m_audio, &a_audio, debug, &min_audio_packet_size);
+ else if (x & AST_FORMAT_VIDEO_MASK)
+ add_vcodec_to_sdp(p, x, &m_video, &a_video, debug, &min_video_packet_size);
else if (x & AST_FORMAT_TEXT_MASK)
- add_tcodec_to_sdp(p, x, 1000,
- &m_text, &a_text, debug, &min_text_packet_size);
+ add_tcodec_to_sdp(p, x, &m_text, &a_text, debug, &min_text_packet_size);
}
/* Now add DTMF RFC2833 telephony-event as a codec */
@@ -9389,7 +9439,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
if (!(p->jointnoncodeccapability & x))
continue;
- add_noncodec_to_sdp(p, x, 8000, &m_audio, &a_audio, debug);
+ add_noncodec_to_sdp(p, x, &m_audio, &a_audio, debug);
}
ast_debug(3, "-- Done with adding codecs to SDP\n");
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-11 21:18:52 +0000