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res_rtp_asterisk: use correct sample rate lookup to account for g722

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Swap out ast_rtp_get_rate for ast_format_get_sample_rate when looking
at the paired audio codec rate to account for g722 oddness.

Resolves: #1657
This commit is contained in:
Mike Bradeen
2026-02-23 16:10:46 -07:00
committed by Asterisk Development Team
parent 5d506fe9b5
commit 09e87f5a15
1 changed files with 6 additions and 2 deletions
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res/res_rtp_asterisk.c
+6 -2
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@@ -4391,15 +4391,19 @@ static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
return -1;
}
/* g722 is a 16K codec that masquerades as an 8K codec within RTP. ast_rtp_get_rate was written specifically to
handle this. If we use the actual sample rate of g722 in this scenario and there is a 16K telephone-event on
offer, we will end up using that instead of the 8K rate telephone-event that is expected with g722. */
if (rtp->lasttxformat == ast_format_none) {
/* No audio frames have been written yet so we have to lookup both the preferred payload type and bitrate. */
payload_format = ast_rtp_codecs_get_preferred_format(ast_rtp_instance_get_codecs(instance));
if (payload_format) {
/* If we have a preferred type, use that. Otherwise default to 8K. */
sample_rate = ast_format_get_sample_rate(payload_format);
sample_rate = ast_rtp_get_rate(payload_format);
}
} else {
sample_rate = ast_format_get_sample_rate(rtp->lasttxformat);
sample_rate = ast_rtp_get_rate(rtp->lasttxformat);
}
if (sample_rate != -1) {