res_pjsip_sdp_rtp: Fix accidentally native bridging calls
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:
parent
efcc6d6f6b
commit
cd793c7c81
|
@ -376,13 +376,16 @@ 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);
|
||||
}
|
||||
}
|
||||
|
||||
|
|
Loading…
Reference in New Issue