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res_pjsip_sdp_rtp: Fix accidentally native bridging calls

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Stop advertising RFC2833 support on the rtp_engine when DTMF mode is
auto but no tel_event was found inside SDP file.

On an incoming call create_rtp will be called and when session->dtmf is
set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without
looking at the SDP file.

Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND
but continued to advertise RFC2833 support.

This meant the native_rtp bridge would falsely consider the two channels
as compatible. In addition to changing the DTMF mode we now set or
remove the AST_RTP_PROPERTY_DTMF.

The property is checked in ast_rtp_dtmf_compatible and called by
native_rtp_bridge_compatible.

ASTERISK-29051 #close

Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287
This commit is contained in:
Holger Hans Peter Freyther
2020-09-23 11:39:12 +08:00
committed by George Joseph
parent 990c72bbcf
commit 9c0ded6e76
1 changed files with 3 additions and 0 deletions
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3
res/res_pjsip_sdp_rtp.c
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@@ -378,13 +378,16 @@ 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_INFO) {
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_NONE);
ast_rtp_instance_set_prop(session_media->rtp, AST_RTP_PROPERTY_DTMF, 0);
}
}