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Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c65
1 files changed, 27 insertions, 38 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index b28c0eed9..e006f06f9 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10175,7 +10175,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
int found = FALSE;
int codec;
char mimeSubtype[128];
- char fmtp_string[64];
+ char fmtp_string[256];
unsigned int sample_rate;
int debug = sip_debug_test_pvt(p);
@@ -10222,12 +10222,17 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
if (debug)
ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
}
- } else if (sscanf(a, "fmtp: %30u %63s", &codec, fmtp_string) == 2) {
+ } else if (sscanf(a, "fmtp: %30u %255s", &codec, fmtp_string) == 2) {
struct ast_format *format;
if ((format = ast_rtp_codecs_get_payload_format(newaudiortp, codec))) {
unsigned int bit_rate;
- int val = 0;
+
+ if (!ast_format_sdp_parse(format, fmtp_string)) {
+ found = TRUE;
+ } else {
+ ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
+ }
switch ((int) format->id) {
case AST_FORMAT_SIREN7:
@@ -10260,21 +10265,6 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
}
}
break;
- case AST_FORMAT_CELT:
- if (sscanf(fmtp_string, "framesize=%30u", &val) == 1) {
- ast_format_append(format, CELT_ATTR_KEY_FRAME_SIZE, val, AST_FORMAT_ATTR_END);
- }
- case AST_FORMAT_SILK:
- if (sscanf(fmtp_string, "maxaveragebitrate=%30u", &val) == 1) {
- ast_format_append(format, SILK_ATTR_KEY_MAX_BITRATE, val, AST_FORMAT_ATTR_END);
- }
- if (sscanf(fmtp_string, "usedtx=%30u", &val) == 1) {
- ast_format_append(format, SILK_ATTR_KEY_DTX, val ? 1 : 0, AST_FORMAT_ATTR_END);
- }
- if (sscanf(fmtp_string, "useinbandfec=%30u", &val) == 1) {
- ast_format_append(format, SILK_ATTR_KEY_FEC, val ? 1 : 0, AST_FORMAT_ATTR_END);
- }
- break;
}
}
}
@@ -10289,6 +10279,7 @@ static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_
char mimeSubtype[128];
unsigned int sample_rate;
int debug = sip_debug_test_pvt(p);
+ char fmtp_string[256];
if (sscanf(a, "rtpmap: %30u %127[^/]/%30u", &codec, mimeSubtype, &sample_rate) == 3) {
/* We have a rtpmap to handle */
@@ -10311,6 +10302,16 @@ static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_
if (debug)
ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
}
+ } else if (sscanf(a, "fmtp: %30u %255s", &codec, fmtp_string) == 2) {
+ struct ast_format *format;
+
+ if ((format = ast_rtp_codecs_get_payload_format(newvideortp, codec))) {
+ if (!ast_format_sdp_parse(format, fmtp_string)) {
+ found = TRUE;
+ } else {
+ ast_rtp_codecs_payloads_unset(newvideortp, NULL, codec);
+ }
+ }
}
return found;
@@ -11747,7 +11748,6 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
{
int rtp_code;
struct ast_format_list fmt;
- int val = 0;
if (debug)
ast_verbose("Adding codec %d (%s) to SDP\n", format->id, ast_getformatname(format));
@@ -11765,6 +11765,8 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
ast_rtp_lookup_mime_subtype2(1, format, 0, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0),
ast_rtp_lookup_sample_rate2(1, format, 0));
+ ast_format_sdp_generate(format, rtp_code, a_buf);
+
switch ((int) format->id) {
case AST_FORMAT_G729A:
/* Indicate that we don't support VAD (G.729 annex B) */
@@ -11790,22 +11792,6 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
/* Indicate that we only expect 64Kbps */
ast_str_append(a_buf, 0, "a=fmtp:%d bitrate=64000\r\n", rtp_code);
break;
- case AST_FORMAT_CELT:
- if (!ast_format_get_value(format, CELT_ATTR_KEY_FRAME_SIZE, &val) && val > 0) {
- ast_str_append(a_buf, 0, "a=fmtp:%d framesize=%u\r\n", rtp_code, val);
- }
- break;
- case AST_FORMAT_SILK:
- if (!ast_format_get_value(format, SILK_ATTR_KEY_MAX_BITRATE, &val) && val > 5000 && val < 40000) {
- ast_str_append(a_buf, 0, "a=fmtp:%d maxaveragebitrate=%u\r\n", rtp_code, val);
- }
- if (!ast_format_get_value(format, SILK_ATTR_KEY_DTX, &val)) {
- ast_str_append(a_buf, 0, "a=fmtp:%d usedtx=%u\r\n", rtp_code, val ? 1 : 0);
- }
- if (!ast_format_get_value(format, SILK_ATTR_KEY_FEC, &val)) {
- ast_str_append(a_buf, 0, "a=fmtp:%d useinbandfec=%u\r\n", rtp_code, val ? 1 : 0);
- }
- break;
}
if (fmt.cur_ms && (fmt.cur_ms < *min_packet_size))
@@ -11837,7 +11823,8 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_rtp_lookup_mime_subtype2(1, format, 0, 0),
ast_rtp_lookup_sample_rate2(1, format, 0));
- /* Add fmtp code here */
+
+ ast_format_sdp_generate(format, rtp_code, a_buf);
}
/*! \brief Add text codec offer to SDP offer/answer body in INVITE or 200 OK */
@@ -12262,10 +12249,12 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
/* Start by sending our preferred audio/video codecs */
for (x = 0; x < AST_CODEC_PREF_SIZE; x++) {
- if (!(ast_codec_pref_index(&p->prefs, x, &tmp_fmt)))
+ struct ast_format pref;
+
+ if (!(ast_codec_pref_index(&p->prefs, x, &pref)))
break;
- if (!(ast_format_cap_iscompatible(tmpcap, &tmp_fmt)))
+ if (!ast_format_cap_get_compatible_format(tmpcap, &pref, &tmp_fmt))
continue;
if (ast_format_cap_iscompatible(alreadysent, &tmp_fmt))
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-21 16:25:16 +0000