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res_pjsip_sdp_rtp.c: Fix DTMF Handling in Re-INVITE with dtmf_mode set to auto

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When the endpoint dtmf_mode is set to auto, a SIP request is sent to the UAC, and the SIP SDP from the UAC does not include the telephone-event. Later, the UAC sends an INVITE, and the SIP SDP includes the telephone-event. In this case, DTMF should be sent by RFC2833 rather than using inband signaling.

Resolves: asterisk#826
(cherry picked from commit 6cf6856080)
This commit is contained in:
Tinet-mucw
2024-08-02 16:49:58 +08:00
committed by Asterisk Development Team
parent 894e509b08
commit 84e93d1b57
1 changed files with 8 additions and 3 deletions
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11
res/res_pjsip_sdp_rtp.c
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@@ -384,9 +384,14 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
}
}
if (!tel_event && (session->dtmf == AST_SIP_DTMF_AUTO)) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 0);
if (session->dtmf == AST_SIP_DTMF_AUTO) {
if (tel_event) {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_RFC2833);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 1);
} else {
ast_rtp_instance_dtmf_mode_set(session_media->rtp, AST_RTP_DTMF_MODE_INBAND);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 0);
}
}
if (session->dtmf == AST_SIP_DTMF_AUTO_INFO) {