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res_pjsip_sdp_rtp: Add support for default/mismatched 8K RFC 4733/2833 digits

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After change made in 624f509 to add support for non 8K RFC 4733/2833 digits,
Asterisk would only accept RFC 4733/2833 offers that matched the sample rate of
the negotiated codec(s).

This change allows Asterisk to accept 8K RFC 4733/2833 offers if the UAC
offfers 8K RFC 4733/2833 but negotiates for a non 8K bitrate codec.

A number of corresponding tests in tests/channels/pjsip/dtmf_sdp also needed to
be re-written to allow for these scenarios.

Fixes: #776
(cherry picked from commit 7d53986262)
This commit is contained in:
Mike Bradeen
2024-06-21 16:56:11 -06:00
committed by Asterisk Development Team
parent 31ff20988c
commit 56f1c20952
3 changed files with 33 additions and 7 deletions
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10
res/res_rtp_asterisk.c
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@@ -4326,11 +4326,17 @@ static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
/* 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 this returns -1, we are being asked to send digits for a sample rate that is outside
what was negotiated for. Fall back if possible. */
/* 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) {
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. */
if (payload == -1) {
return -1;
}
ast_test_suite_event_notify("DTMF_BEGIN", "Digit: %d\r\nPayload: %d\r\nRate: %d\r\n", digit, payload, sample_rate);
ast_debug(1, "Sending digit '%d' at rate %d with payload %d\n", digit, sample_rate, payload);