From 4499fbc81964a22e93676eb55801e9abe4e3ccd0 Mon Sep 17 00:00:00 2001 From: Holger Hans Peter Freyther <holger@moiji-mobile.com> Date: Wed, 23 Sep 2020 11:39:12 +0800 Subject: [PATCH] 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 --- res/res_pjsip_sdp_rtp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/res/res_pjsip_sdp_rtp.c b/res/res_pjsip_sdp_rtp.c index eacae22dd6..1bccc504b2 100644 --- a/res/res_pjsip_sdp_rtp.c +++ b/res/res_pjsip_sdp_rtp.c @@ -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); } } -- GitLab