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FS-5257 --resolve you seem to be setting sip_secure_media on an inbound leg, this is not correct. This patch should auto-correct that but also beware that the variable has changed to rtp_secure_media after this commit

This commit is contained in:
Anthony Minessale 2013-04-02 20:05:46 -05:00
parent 5544db8d5e
commit 69c3c7d8e2
11 changed files with 37 additions and 22 deletions
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6
conf/insideout/dialplan/default.xml
View File

@ -95,9 +95,9 @@
<action application="info"/>
</condition>
<condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
<action application="set" data="sip_secure_media=true"/>
<action application="set" data="rtp_secure_media=true"/>
<!-- Offer SRTP on outbound legs if we have it on inbound. -->
<!-- <action application="export" data="sip_secure_media=true"/> -->
<!-- <action application="export" data="rtp_secure_media=true"/> -->
</condition>
<condition>
<action application="db" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
@ -231,7 +231,7 @@
<anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
<!--<anti-action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
<!--<anti-action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
<anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
<anti-action application="answer"/>

2
conf/insideout/dialplan/features.xml
View File

@ -40,7 +40,7 @@
<extension name="is_secure">
<!-- Only Truly consider it secure if its TLS and SRTP -->
<condition field="${sip_via_protocol}" expression="tls"/>
<condition field="${sip_secure_media_confirmed}" expression="^true$">
<condition field="${rtp_secure_media_confirmed}" expression="^true$">
<action application="sleep" data="1000"/>
<action application="playback" data="misc/call_secured.wav"/>
<anti-action application="eval" data="not_secure"/>

4
conf/insideout/directory/default/brian.xml
View File

@ -60,8 +60,8 @@
<variable name="effective_caller_id_number" value="1000"/>
<!-- Don't write a CDR if this is false valid values are: true, false, a_leg and b_leg -->
<variable name="process_cdr" value="true"/>
<!-- sip_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
<variable name="sip_secure_media" value="true"/>
<!-- rtp_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
<variable name="rtp_secure_media" value="true"/>
<!-- limit the max number of outgoing calls for this user -->
<!--<variable name="max_calls" value="2"/>-->

6
conf/sbc/dialplan/default.xml
View File

@ -95,9 +95,9 @@
<action application="info"/>
</condition>
<condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
<action application="set" data="sip_secure_media=true"/>
<action application="set" data="rtp_secure_media=true"/>
<!-- Offer SRTP on outbound legs if we have it on inbound. -->
<!-- <action application="export" data="sip_secure_media=true"/> -->
<!-- <action application="export" data="rtp_secure_media=true"/> -->
</condition>
<condition>
<action application="db" data="insert/${domain_name}-spymap/${caller_id_number}/${uuid}"/>
@ -231,7 +231,7 @@
<anti-action application="db" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
<anti-action application="db" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
<anti-action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
<!--<anti-action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
<!--<anti-action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
<anti-action application="db" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<anti-action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
<anti-action application="answer"/>

2
conf/sbc/dialplan/features.xml
View File

@ -40,7 +40,7 @@
<extension name="is_secure">
<!-- Only Truly consider it secure if its TLS and SRTP -->
<condition field="${sip_via_protocol}" expression="tls"/>
<condition field="${sip_secure_media_confirmed}" expression="^true$">
<condition field="${rtp_secure_media_confirmed}" expression="^true$">
<action application="sleep" data="1000"/>
<action application="playback" data="misc/call_secured.wav"/>
<anti-action application="eval" data="not_secure"/>

10
conf/vanilla/dialplan/default.xml
View File

@ -136,9 +136,9 @@
</condition>
-->
<condition field="${rtp_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never">
<action application="set" data="sip_secure_media=true"/>
<action application="set" data="rtp_secure_media=true"/>
<!-- Offer SRTP on outbound legs if we have it on inbound. -->
<!-- <action application="export" data="sip_secure_media=true"/> -->
<!-- <action application="export" data="rtp_secure_media=true"/> -->
</condition>
<!--
@ -147,9 +147,9 @@
-->
<condition field="${endpoint_disposition}" expression="^(DELAYED NEGOTIATION)"/>
<condition field="${switch_r_sdp}" expression="(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)" break="never">
<action application="set" data="sip_secure_media=true"/>
<action application="set" data="rtp_secure_media=true"/>
<!-- Offer SRTP on outbound legs if we have it on inbound. -->
<!-- <action application="export" data="sip_secure_media=true"/> -->
<!-- <action application="export" data="rtp_secure_media=true"/> -->
</condition>
@ -274,7 +274,7 @@
<action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/>
<action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>
<!--<action application="export" data="nolocal:sip_secure_media=${user_data(${dialed_extension}@${domain_name} var sip_secure_media)}"/>-->
<!--<action application="export" data="nolocal:rtp_secure_media=${user_data(${dialed_extension}@${domain_name} var rtp_secure_media)}"/>-->
<action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>
<action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
<action application="answer"/>

2
conf/vanilla/dialplan/features.xml
View File

@ -56,7 +56,7 @@
<extension name="is_secure" continue="true">
<!-- Only Truly consider it secure if its TLS and SRTP -->
<condition field="${sip_via_protocol}" expression="tls"/>
<condition field="${sip_secure_media_confirmed}" expression="^true$">
<condition field="${rtp_secure_media_confirmed}" expression="^true$">
<action application="sleep" data="1000"/>
<action application="playback" data="misc/call_secured.wav"/>
<anti-action application="eval" data="not_secure"/>

4
conf/vanilla/directory/default/brian.xml
View File

@ -62,8 +62,8 @@
<variable name="effective_caller_id_number" value="1000"/>
<!-- Don't write a CDR if this is false valid values are: true, false, a_leg and b_leg -->
<variable name="process_cdr" value="true"/>
<!-- sip_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
<variable name="sip_secure_media" value="true"/>
<!-- rtp_secure_media will offer mandatory SRTP on invite AES_CM_128_HMAC_SHA1_32, AES_CM_128_HMAC_SHA1_80 or true-->
<variable name="rtp_secure_media" value="true"/>
<!-- limit the max number of outgoing calls for this user -->
<!--<variable name="max_calls" value="2"/>-->

4
src/mod/endpoints/mod_sofia/mod_sofia.h
View File

@ -112,8 +112,8 @@ typedef struct private_object private_object_t;
#define SOFIA_DEFAULT_PORT "5060"
#define SOFIA_DEFAULT_TLS_PORT "5061"
#define SOFIA_REFER_TO_VARIABLE "sip_refer_to"
#define SOFIA_SECURE_MEDIA_VARIABLE "sip_secure_media"
#define SOFIA_SECURE_MEDIA_CONFIRMED_VARIABLE "sip_secure_media_confirmed"
#define SOFIA_SECURE_MEDIA_VARIABLE "rtp_secure_media"
#define SOFIA_SECURE_MEDIA_CONFIRMED_VARIABLE "rtp_secure_media_confirmed"
#define SOFIA_SECURE_VIDEO_CONFIRMED_VARIABLE "sip_secure_video_confirmed"
//#define SOFIA_HAS_CRYPTO_VARIABLE "rtp_has_crypto"
//#define SOFIA_HAS_VIDEO_CRYPTO_VARIABLE "sip_has_video_crypto"

17
src/switch_channel.c
View File

@ -3155,8 +3155,20 @@ SWITCH_DECLARE(void) switch_channel_check_zrtp(switch_channel_t *channel)
}
}
static void check_secure(switch_channel_t *channel)
{
const char *var, *sec;
if (switch_channel_direction(channel) == SWITCH_CALL_DIRECTION_INBOUND) {
if ((sec = switch_channel_get_variable(channel, "rtp_secure_media")) && switch_true(sec)) {
if (!(var = switch_channel_get_variable(channel, "rtp_has_crypto"))) {
switch_log_printf(SWITCH_CHANNEL_CHANNEL_LOG(channel), SWITCH_LOG_WARNING, "rtp_secure_media invalid in this context.\n");
switch_channel_set_variable(channel, "rtp_secure_media", NULL);
}
}
}
}
SWITCH_DECLARE(switch_status_t) switch_channel_perform_mark_pre_answered(switch_channel_t *channel, const char *file, const char *func, int line)
{
@ -3241,6 +3253,8 @@ SWITCH_DECLARE(switch_status_t) switch_channel_perform_pre_answer(switch_channel
return SWITCH_STATUS_SUCCESS;
}
check_secure(channel);
if (switch_channel_direction(channel) == SWITCH_CALL_DIRECTION_INBOUND) {
msg.message_id = SWITCH_MESSAGE_INDICATE_PROGRESS;
msg.from = channel->name;
@ -3511,7 +3525,8 @@ SWITCH_DECLARE(switch_status_t) switch_channel_perform_answer(switch_channel_t *
if (switch_channel_test_flag(channel, CF_ANSWERED)) {
return SWITCH_STATUS_SUCCESS;
}
check_secure(channel);
msg.message_id = SWITCH_MESSAGE_INDICATE_ANSWER;
msg.from = channel->name;

2
src/switch_rtp.c
View File

@ -2656,7 +2656,7 @@ SWITCH_DECLARE(switch_status_t) switch_rtp_create(switch_rtp_t **new_rtp_session
int initiator = 0;
const char *zrtp_enabled = switch_channel_get_variable(channel, "zrtp_secure_media");
const char *srtp_enabled = switch_channel_get_variable(channel, "sip_secure_media");
const char *srtp_enabled = switch_channel_get_variable(channel, "rtp_secure_media");
if (switch_true(srtp_enabled) && switch_true(zrtp_enabled)) {
switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_WARNING,