Allow early media (SDP in 1xx messages) to change the formats list.
git-svn-id: http://voip.null.ro/svn/yate@1979 acf43c95-373e-0410-b603-e72c3f656dc1
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@ -173,3 +173,6 @@
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; ignore_missing_ack: bool: Ignore missing ACK on INVITE, don't drop the calls
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;ignore_missing_ack=no
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; 1xx_change_formats: bool: Provisional messages can change the formats list
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;1xx_change_formats=yes
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@ -576,6 +576,7 @@ static bool s_rfc2833 = true;
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static bool s_forward_sdp = false;
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static bool s_start_rtp = false;
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static bool s_ack_required = true;
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static bool s_1xx_formats = true;
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static bool s_auth_register = true;
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static bool s_multi_ringing = false;
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static bool s_refresh_nosdp = true;
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@ -2406,6 +2407,8 @@ MimeSdpBody* YateSIPConnection::createProvisionalSDP(Message& msg)
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return 0;
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if (m_rtpAddr.null())
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return 0;
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if (s_1xx_formats)
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updateFormats(msg);
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return createRtpSDP(true);
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}
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@ -4517,6 +4520,7 @@ void SIPDriver::initialize()
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s_auth_register = s_cfg.getBoolValue("registrar","auth_required",true);
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s_nat_refresh = s_cfg.getIntValue("registrar","nat_refresh",25);
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s_ack_required = !s_cfg.getBoolValue("hacks","ignore_missing_ack",false);
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s_1xx_formats = s_cfg.getBoolValue("hacks","1xx_change_formats",true);
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initAudioCodecs();
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if (!m_endpoint) {
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m_endpoint = new YateSIPEndPoint();
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