SRTP calls fail 'because secure media is administratively disabled'
I am using FreeSWITCH demo configuration to test SRTP calls. I am placing a call between endpoint 1000 and 1001. The call works fine when SRTP is disabled on the softphones. The problem starts when I enable SRTP on the softphones. I also enabled SRTP on FreeSWITCH server, by including the following command in vars.xml: <X-PRE-PROCESS cmd="set" data="sip_secure_media=true"/> When I try to place a call between the two endpoints now, I end up with one side immediately hanging up. Specifically, the error seems to be: [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled Followed by: [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] Couldn't find any info on how to resolve this issue online. Here's the relevant log: 2016-03-18 09:45:21.664536 [INFO] switch_rtp.c:3605 Activating Audio Secure RTP SEND [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [INFO] switch_rtp.c:3583 Activating Audio Secure RTP RECV [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@10.10.10.77! [DEBUG] switch_channel.c:3461 (sofia/internal/1000@10.10.10.77) Callstate Change RINGING -> EARLY [DEBUG] mod_sofia.c:2330 Ring SDP: v=0 o=FreeSWITCH 1458288533 1458288534 IN IP4 10.10.10.77 s=FreeSWITCH c=IN IP4 10.10.10.77 t=0 0 m=audio 20188 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhco9Z9FgciE1vh5LQqGyNXxts0etxavW51gJKVP [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@8000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/1000@10.10.10.77 Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6756 Channel sofia/internal/1000@10.10.10.77 entering state [early][183] [DEBUG] switch_rtp.c:6645 Correct audio ip/port confirmed. [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [completing][200] [DEBUG] sofia.c:6766 Remote SDP: v=0 o=VOCAL 61793 12653 IN IP4 10.10.10.112 s=- c=IN IP4 10.10.10.112 t=0 0 m=audio 1442 RTP/SAVP 0 a=rtpmap:0 PCMU/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i7qQOAKGWeBs/QtXQcYRD3g7u2pGv/BDqrvKE6Av a=ptime:20 [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [ready][200] [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] [DEBUG] sofia.c:1355 Channel is already hungup. 2016-03-18 09:45:26.303802
They changed the name to rtp_secure_media sometime in the last few months. Eternal21 <eternal24@gmail.com> wrote:
I am using FreeSWITCH demo configuration to test SRTP calls. I am placing a call between endpoint 1000 and 1001. The call works fine when SRTP is disabled on the softphones. The problem starts when I enable SRTP on the softphones. I also enabled SRTP on FreeSWITCH server, by including the following command in vars.xml:
<X-PRE-PROCESS cmd="set" data="sip_secure_media=true"/>
When I try to place a call between the two endpoints now, I end up with one side immediately hanging up. Specifically, the error seems to be: [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled
Followed by: [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
Couldn't find any info on how to resolve this issue online.
Here's the relevant log: 2016-03-18 09:45:21.664536
[INFO] switch_rtp.c:3605 Activating Audio Secure RTP SEND [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [INFO] switch_rtp.c:3583 Activating Audio Secure RTP RECV [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@10.10.10.77! [DEBUG] switch_channel.c:3461 (sofia/internal/1000@10.10.10.77) Callstate Change RINGING -> EARLY [DEBUG] mod_sofia.c:2330 Ring SDP: v=0 o=FreeSWITCH 1458288533 1458288534 IN IP4 10.10.10.77 s=FreeSWITCH c=IN IP4 10.10.10.77 t=0 0 m=audio 20188 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhco9Z9FgciE1vh5LQqGyNXxts0etxavW51gJKVP
[DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@8000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/1000@10.10.10.77 Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6756 Channel sofia/internal/1000@10.10.10.77 entering state [early][183] [DEBUG] switch_rtp.c:6645 Correct audio ip/port confirmed. [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [completing][200] [DEBUG] sofia.c:6766 Remote SDP: v=0 o=VOCAL 61793 12653 IN IP4 10.10.10.112 s=- c=IN IP4 10.10.10.112 t=0 0 m=audio 1442 RTP/SAVP 0 a=rtpmap:0 PCMU/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i7qQOAKGWeBs/QtXQcYRD3g7u2pGv/BDqrvKE6Av a=ptime:20
[DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [ready][200] [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] [DEBUG] sofia.c:1355 Channel is already hungup.
2016-03-18 09:45:26.303802
---------------------------------------------------- Alternatives:
---------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
Thank you for a prompt response. Replacing sip_secure_media with rtp_secure_media resolved my issue. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of covici@ccs.covici.com Sent: Friday, March 18, 2016 11:05 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] SRTP calls fail 'because secure media is administratively disabled' They changed the name to rtp_secure_media sometime in the last few months. Eternal21 <eternal24@gmail.com> wrote:
I am using FreeSWITCH demo configuration to test SRTP calls. I am placing a call between endpoint 1000 and 1001. The call works fine when SRTP is disabled on the softphones. The problem starts when I enable SRTP on the softphones. I also enabled SRTP on FreeSWITCH server, by including the following command in vars.xml:
<X-PRE-PROCESS cmd="set" data="sip_secure_media=true"/>
When I try to place a call between the two endpoints now, I end up with one side immediately hanging up. Specifically, the error seems to be: [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled
Followed by: [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
Couldn't find any info on how to resolve this issue online.
Here's the relevant log: 2016-03-18 09:45:21.664536
[INFO] switch_rtp.c:3605 Activating Audio Secure RTP SEND [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [INFO] switch_rtp.c:3583 Activating Audio Secure RTP RECV [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@10.10.10.77! [DEBUG] switch_channel.c:3461 (sofia/internal/1000@10.10.10.77) Callstate Change RINGING -> EARLY [DEBUG] mod_sofia.c:2330 Ring SDP: v=0 o=FreeSWITCH 1458288533 1458288534 IN IP4 10.10.10.77 s=FreeSWITCH c=IN IP4 10.10.10.77 t=0 0 m=audio 20188 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhco9Z9FgciE1vh5LQqGyNXxts0etxavW51gJKVP
[DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@8000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/1000@10.10.10.77 Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6756 Channel sofia/internal/1000@10.10.10.77 entering state [early][183] [DEBUG] switch_rtp.c:6645 Correct audio ip/port confirmed. [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [completing][200] [DEBUG] sofia.c:6766 Remote SDP: v=0 o=VOCAL 61793 12653 IN IP4 10.10.10.112 s=- c=IN IP4 10.10.10.112 t=0 0 m=audio 1442 RTP/SAVP 0 a=rtpmap:0 PCMU/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i7qQOAKGWeBs/QtXQcYRD3g7u2pGv/BDqrvKE6Av a=ptime:20
[DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [ready][200] [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] [DEBUG] sofia.c:1355 Channel is already hungup.
2016-03-18 09:45:26.303802
---------------------------------------------------- Alternatives:
---------------------------------------------------- ______________________________________________________________________ ___ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
That change was made between 1.2 and 1.4, its been multiple years now, not recent.
On Mar 18, 2016, at 11:04 AM, covici@ccs.covici.com wrote:
They changed the name to rtp_secure_media sometime in the last few months.
Eternal21 <eternal24@gmail.com <mailto:eternal24@gmail.com>> wrote:
I am using FreeSWITCH demo configuration to test SRTP calls. I am placing a call between endpoint 1000 and 1001. The call works fine when SRTP is disabled on the softphones. The problem starts when I enable SRTP on the softphones. I also enabled SRTP on FreeSWITCH server, by including the following command in vars.xml:
<X-PRE-PROCESS cmd="set" data="sip_secure_media=true"/>
When I try to place a call between the two endpoints now, I end up with one side immediately hanging up. Specifically, the error seems to be: [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled
Followed by: [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
Couldn't find any info on how to resolve this issue online.
Here's the relevant log: 2016-03-18 09:45:21.664536
[INFO] switch_rtp.c:3605 Activating Audio Secure RTP SEND [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [INFO] switch_rtp.c:3583 Activating Audio Secure RTP RECV [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@10.10.10.77! [DEBUG] switch_channel.c:3461 (sofia/internal/1000@10.10.10.77) Callstate Change RINGING -> EARLY [DEBUG] mod_sofia.c:2330 Ring SDP: v=0 o=FreeSWITCH 1458288533 1458288534 IN IP4 10.10.10.77 s=FreeSWITCH c=IN IP4 10.10.10.77 t=0 0 m=audio 20188 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhco9Z9FgciE1vh5LQqGyNXxts0etxavW51gJKVP
[DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@8000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/1000@10.10.10.77 Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6756 Channel sofia/internal/1000@10.10.10.77 entering state [early][183] [DEBUG] switch_rtp.c:6645 Correct audio ip/port confirmed. [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [completing][200] [DEBUG] sofia.c:6766 Remote SDP: v=0 o=VOCAL 61793 12653 IN IP4 10.10.10.112 s=- c=IN IP4 10.10.10.112 t=0 0 m=audio 1442 RTP/SAVP 0 a=rtpmap:0 PCMU/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i7qQOAKGWeBs/QtXQcYRD3g7u2pGv/BDqrvKE6Av a=ptime:20
[DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [ready][200] [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] [DEBUG] sofia.c:1355 Channel is already hungup.
2016-03-18 09:45:26.303802
---------------------------------------------------- Alternatives:
---------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com <mailto:covici@ccs.covici.com>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org/> http://confluence.freeswitch.org <http://confluence.freeswitch.org/> http://www.cluecon.com <http://www.cluecon.com/>
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I was going by examples from FreeSWITCH 1.2 book (didn’t see a later release on Amazon). The following wiki states outdated values as well (and happens to be the first link when googling for freeswitch SRTP): https://wiki.freeswitch.org/wiki/SRTP *From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Michael Jerris *Sent:* Friday, March 18, 2016 2:12 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] SRTP calls fail 'because secure media is administratively disabled' That change was made between 1.2 and 1.4, its been multiple years now, not recent. On Mar 18, 2016, at 11:04 AM, covici@ccs.covici.com wrote: They changed the name to rtp_secure_media sometime in the last few months. Eternal21 <eternal24@gmail.com> wrote: I am using FreeSWITCH demo configuration to test SRTP calls. I am placing a call between endpoint 1000 and 1001. The call works fine when SRTP is disabled on the softphones. The problem starts when I enable SRTP on the softphones. I also enabled SRTP on FreeSWITCH server, by including the following command in vars.xml: <X-PRE-PROCESS cmd="set" data="sip_secure_media=true"/> When I try to place a call between the two endpoints now, I end up with one side immediately hanging up. Specifically, the error seems to be: [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled Followed by: [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] Couldn't find any info on how to resolve this issue online. Here's the relevant log: 2016-03-18 09:45:21.664536 [INFO] switch_rtp.c:3605 Activating Audio Secure RTP SEND [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [INFO] switch_rtp.c:3583 Activating Audio Secure RTP RECV [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@10.10.10.77! [DEBUG] switch_channel.c:3461 (sofia/internal/1000@10.10.10.77) Callstate Change RINGING -> EARLY [DEBUG] mod_sofia.c:2330 Ring SDP: v=0 o=FreeSWITCH 1458288533 1458288534 IN IP4 10.10.10.77 s=FreeSWITCH c=IN IP4 10.10.10.77 t=0 0 m=audio 20188 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhco9Z9FgciE1vh5LQqGyNXxts0etxavW51gJKVP [DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@8000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/1000@10.10.10.77 Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6756 Channel sofia/internal/1000@10.10.10.77 entering state [early][183] [DEBUG] switch_rtp.c:6645 Correct audio ip/port confirmed. [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [completing][200] [DEBUG] sofia.c:6766 Remote SDP: v=0 o=VOCAL 61793 12653 IN IP4 10.10.10.112 s=- c=IN IP4 10.10.10.112 t=0 0 m=audio 1442 RTP/SAVP 0 a=rtpmap:0 PCMU/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i7qQOAKGWeBs/QtXQcYRD3g7u2pGv/BDqrvKE6Av a=ptime:20 [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [ready][200] [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] [DEBUG] sofia.c:1355 Channel is already hungup. 2016-03-18 09:45:26.303802 ---------------------------------------------------- Alternatives: ---------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
On Fri, Mar 18, 2016 at 11:21 AM, Piotr Starzyk < pstarzyk@general-devices.com> wrote:
I was going by examples from FreeSWITCH 1.2 book (didn’t see a later release on Amazon).
The following wiki states outdated values as well (and happens to be the first link when googling for freeswitch SRTP):
The docs are all being migrated over to Confluence from the old wiki. Any time you see a link to something on wiki.freeswitch.org you should definitely check the Confluence <https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+Explained> page first. In a pinch just click the "Docs" link on freeswitch.org. That all being said, there does not seem to be an equivalent page in Confluence. However, if you search for "sip_secure_media" in Confluence you'll see several links that mention the change to "rtp_secure_media" and other helpful information. -MSC
Think you mean YEAR On Friday, March 18, 2016, <covici@ccs.covici.com> wrote:
They changed the name to rtp_secure_media sometime in the last few months.
Eternal21 <eternal24@gmail.com <javascript:;>> wrote:
I am using FreeSWITCH demo configuration to test SRTP calls. I am placing a call between endpoint 1000 and 1001. The call works fine when SRTP is disabled on the softphones. The problem starts when I enable SRTP on the softphones. I also enabled SRTP on FreeSWITCH server, by including the following command in vars.xml:
<X-PRE-PROCESS cmd="set" data="sip_secure_media=true"/>
When I try to place a call between the two endpoints now, I end up with one side immediately hanging up. Specifically, the error seems to be: [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled
Followed by: [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
Couldn't find any info on how to resolve this issue online.
Here's the relevant log: 2016-03-18 09:45:21.664536
[INFO] switch_rtp.c:3605 Activating Audio Secure RTP SEND [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [INFO] switch_rtp.c:3583 Activating Audio Secure RTP RECV [DEBUG] switch_core_sqldb.c:2607 Secure Type: srtp:sdes:AES_CM_128_HMAC_SHA1_80 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@10.10.10.77 <javascript:;>! [DEBUG] switch_channel.c:3461 (sofia/internal/1000@10.10.10.77 <javascript:;>) Callstate Change RINGING -> EARLY [DEBUG] mod_sofia.c:2330 Ring SDP: v=0 o=FreeSWITCH 1458288533 1458288534 IN IP4 10.10.10.77 s=FreeSWITCH c=IN IP4 10.10.10.77 t=0 0 m=audio 20188 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhco9Z9FgciE1vh5LQqGyNXxts0etxavW51gJKVP
[DEBUG] switch_ivr_originate.c:1274 Raw Codec Activation Success L16@8000hz 1 channel 20ms [DEBUG] switch_core_codec.c:221 sofia/internal/1000@10.10.10.77 <javascript:;> Push codec L16:100 [DEBUG] switch_ivr_originate.c:1343 Play Ringback Tone [%(2000,4000,440,480)] [DEBUG] sofia.c:6756 Channel sofia/internal/1000@10.10.10.77 <javascript:;> entering state [early][183] [DEBUG] switch_rtp.c:6645 Correct audio ip/port confirmed. [DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [completing][200] [DEBUG] sofia.c:6766 Remote SDP: v=0 o=VOCAL 61793 12653 IN IP4 10.10.10.112 s=- c=IN IP4 10.10.10.112 t=0 0 m=audio 1442 RTP/SAVP 0 a=rtpmap:0 PCMU/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:i7qQOAKGWeBs/QtXQcYRD3g7u2pGv/BDqrvKE6Av a=ptime:20
[DEBUG] sofia.c:6756 Channel sofia/internal/1001@10.10.10.112:5060 entering state [ready][200] [WARNING] switch_core_media.c:4035 Declining invite with only SAVP because secure media is administratively disabled [NOTICE] sofia.c:7742 Hangup sofia/internal/1001@10.10.10.112:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] [DEBUG] sofia.c:1355 Channel is already hungup.
2016-03-18 09:45:26.303802
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John Covici covici@ccs.covici.com <javascript:;>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:;> http://www.freeswitchsolutions.com
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participants (6)
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Brian West
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covici＠ccs.covici.com
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Eternal21
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Michael Collins
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Michael Jerris
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Piotr Starzyk