kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-12 15:45:18 +00:00
Code Issues Packages Projects Releases Wiki Activity

res_pjsip_sdp_rtp: Use negotiated DTMF Payload types on bitrate mismatch

Browse Source 
When Asterisk sends an offer to Bob that includes 48K and 8K codecs with
matching 4733 offers, Bob may want to use the 48K audio codec but can not
accept 48K digits and so negotiates for a mixed set.

Asterisk will now check Bob's offer to make sure Bob has indicated this is
acceptible and if not, will use Bob's preference.

Fixes: #847
This commit is contained in:
Mike Bradeen
2024-08-21 10:11:31 -06:00
parent 6e47295800
commit a8567b92f0
4 changed files with 159 additions and 9 deletions
Download Patch File Download Diff File Expand all files Collapse all files

25
res/res_rtp_asterisk.c
View File

@@ -4283,8 +4283,7 @@ static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
{
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
struct ast_sockaddr remote_address = { {0,} };
int hdrlen = 12, res = 0, i = 0, payload = 101;
unsigned int sample_rate = 8000;
int hdrlen = 12, res = 0, i = 0, payload = -1, sample_rate = -1;
char data[256];
unsigned int *rtpheader = (unsigned int*)data;
RAII_VAR(struct ast_format *, payload_format, NULL, ao2_cleanup);
@@ -4323,16 +4322,26 @@ static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
sample_rate = ast_format_get_sample_rate(rtp->lasttxformat);
}
/* Grab the matching DTMF type payload */
payload = ast_rtp_codecs_payload_code_tx_sample_rate(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_DTMF, sample_rate);
if (sample_rate != -1) {
payload = ast_rtp_codecs_payload_code_tx_sample_rate(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_DTMF, sample_rate);
}
/* If this returns -1, we are using a codec with a sample rate that does not have a matching RFC 2833/4733
offer. The offer may have included a default-rate one that doesn't match the codec rate, so try to use that. */
if (payload == -1) {
if (payload == -1 ||
!ast_rtp_payload_mapping_tx_is_present(
ast_rtp_instance_get_codecs(instance), ast_rtp_codecs_get_payload(ast_rtp_instance_get_codecs(instance), payload))) {
/* Fall back to the preferred DTMF payload type and sample rate as either we couldn't find an audio codec to try and match
sample rates with or we could, but a telephone-event matching that audio codec's sample rate was not included in the
sdp negotiated by the far end. */
payload = ast_rtp_codecs_get_preferred_dtmf_format_pt(ast_rtp_instance_get_codecs(instance));
sample_rate = ast_rtp_codecs_get_preferred_dtmf_format_rate(ast_rtp_instance_get_codecs(instance));
}
/* The sdp negotiation has not yeilded a usable RFC 2833/4733 format. Try a default-rate one as a last resort. */
if (payload == -1 || sample_rate == -1) {
sample_rate = DEFAULT_DTMF_SAMPLE_RATE_MS;
payload = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_DTMF);
}
/* No default-rate offer either, trying to send a digit outside of what was negotiated for. */
/* Even trying a default payload has failed. We are trying to send a digit outside of what was negotiated for. */
if (payload == -1) {
return -1;
}