SRTP - Can't provide secure audio requested in SDP offer

[noy_siam]

System:
Ubuntu 12.04 LTS 64Bit
Asterisk 11.2.1
FreePBX 2.11

Hi Guys,
I have a problem to issue calls with SRTP.
The FreePBX configured to work with TLS and libSRTP installed well. I used this tutorial to configuration: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
When I am initiating new call with TLS only the calls working well, when SRTP added and initiate new call, I received this error:

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
[2013-04-21 10:39:00] WARNING[16542][C-00000352]: sip/sdp_crypto.c:170 sdp_crypto_activate: Could not set SRTP policies
[2013-04-21 10:39:00] WARNING[16542][C-00000352]: sip/sdp_crypto.c:170 sdp_crypto_activate: Could not set SRTP policies
[2013-04-21 10:39:00] WARNING[16542][C-00000352]: chan_sip.c:10427 process_sdp: Can’t provide secure audio requested in SDP offer

When I tried to Google it, I could not find any relevant posts to solve this issue.
If more information needed, just let me know and I will upload it.

Thanks,
Noy

[david55]

You will need to provide copies of the SDP exchanged (e.g. edited from “sip set debug on” output). You may also need to provide verbose logging from when you do that.

[noy_siam]

Hi David,

Thank you for your replayed, following the requested data:

[2013-04-22 07:07:10] DEBUG[2936]: logger.c:1294 ast_create_callid: CALL_ID [C-000003c1] created by thread. [2013-04-22 07:07:10] DEBUG[2936]: acl.c:979 ast_ouraddrfor: For destination '84.111.xx.xx', our source address is '173.203.xx.xx'. [2013-04-22 07:07:10] DEBUG[2936]: chan_sip.c:4021 ast_sip_ouraddrfor: Setting SIP_TRANSPORT_TLS with address 173.203.xx.xx:5061 [2013-04-22 07:07:10] DEBUG[2936]: chan_sip.c:8721 sip_alloc: Allocating new SIP dialog for dc49a991f8404a278310212a1ecf37da - INVITE (No RTP) [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-000003c1] bound to thread. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:27866 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/reqresp_parser.c:1550 parse_sip_options: Begin: parsing SIP "Supported: 100rel, replaces, norefersub, gruu" [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/reqresp_parser.c:1566 parse_sip_options: Found SIP option: -100rel- [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/reqresp_parser.c:1574 parse_sip_options: Matched SIP option: 100rel [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/reqresp_parser.c:1566 parse_sip_options: Found SIP option: -replaces- [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/reqresp_parser.c:1574 parse_sip_options: Matched SIP option: replaces [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/reqresp_parser.c:1566 parse_sip_options: Found SIP option: -norefersub- [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/reqresp_parser.c:1574 parse_sip_options: Matched SIP option: norefersub [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/reqresp_parser.c:1566 parse_sip_options: Found SIP option: -gruu- [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/reqresp_parser.c:1574 parse_sip_options: Matched SIP option: gruu [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:17845 check_via: NAT detected for 192.168.xx.xx:56511 / 84.111.xx.xx:56511 [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 401' onto TLS socket destined for 84.111.xx.xx:56511 [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-000003c1] being removed from thread. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-000003c1] bound to thread. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:27866 handle_incoming: **** Received ACK (6) - Command in SIP ACK [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:4556 __sip_ack: Stopping retransmission on 'dc49a991f8404a278310212a1ecf37da' of Response 26651: Match Not Found [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-000003c1] being removed from thread. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-000003c1] bound to thread. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:27866 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:17845 check_via: NAT detected for 192.168.xx.xx:56511 / 84.111.xx.xx:56511 [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: rtp_engine.c:283 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7fcf3002a588' [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: res_rtp_asterisk.c:1732 ast_rtp_new: Allocated port 18996 for RTP instance '0x7fcf3002a588' [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: rtp_engine.c:292 ast_rtp_instance_new: RTP instance '0x7fcf3002a588' is setup and ready to go [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fcf3002a588' == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:5725 do_setnat: Setting NAT on RTP to On [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=- 3575614014 3575614014 IN IP4 192.168.xx.xx... OK. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Blink 0.2.10 (Windows)... UNSUPPORTED OR FAILED. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 192.168.xx.xx... OK. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7fcf484d9630 [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fcf484d9630 [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fcf484d9630 [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtcp:50003... UNSUPPORTED OR FAILED. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED OR FAILED. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: sip/sdp_crypto.c:105 sdp_crypto_setup: local_key64 oGykE6dy9DdsfiA7yepYNnVgro4nD5XHzTCPTVHS len 40 [2013-04-22 07:07:10] WARNING[2936][C-000003c1]: sip/sdp_crypto.c:170 sdp_crypto_activate: Could not set SRTP policies [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:UU+lQWE36NuanMUCKV9MiIQoymm6ZmGVAktbBN26... UNSUPPORTED OR FAILED. [2013-04-22 07:07:10] WARNING[2936][C-000003c1]: sip/sdp_crypto.c:170 sdp_crypto_activate: Could not set SRTP policies [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:swhB7TrakQOyuYC19qc2u9is2RDEaYGC3FvSEV3i... UNSUPPORTED OR FAILED. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=sendrecv... OK. [2013-04-22 07:07:10] WARNING[2936][C-000003c1]: chan_sip.c:10427 process_sdp: Can't provide secure audio requested in SDP offer [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 488' onto TLS socket destined for 84.111.xx.xx:56511 [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:25116 handle_request_invite: No compatible codecs for this SIP call. [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: chan_sip.c:28168 handle_request_do: SIP message could not be handled, bad request: dc49a991f8404a278310212a1ecf37da [2013-04-22 07:07:10] DEBUG[2936][C-000003c1]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-000003c1] being removed from thread. [2013-04-22 07:07:11] DEBUG[2936][C-000003c1]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-000003c1] bound to thread. [2013-04-22 07:07:11] DEBUG[2936][C-000003c1]: chan_sip.c:27866 handle_incoming: **** Received ACK (6) - Command in SIP ACK [2013-04-22 07:07:11] DEBUG[2936][C-000003c1]: chan_sip.c:4556 __sip_ack: Stopping retransmission on 'dc49a991f8404a278310212a1ecf37da' of Response 26652: Match Not Found [2013-04-22 07:07:11] DEBUG[2936][C-000003c1]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-000003c1] being removed from thread.

And…

[code]<— SIP read from TLS:84.111.xx.xx:56511 —>
INVITE sip:*97@173.203.xx.xx SIP/2.0
Via: SIP/2.0/tls 192.168.xx.xx:56511;rport;branch=z9hG4bKPj1b3f7663735d448a9e6c9f153ccffe67
Max-Forwards: 70
From: “20000” sip:20000@173.203.xx.xx;tag=ac06ae02257c4b20b20dda73e286dc4e
To: sip:*97@173.203.xx.xx
Contact: sip:aeljrvms@192.168.xx.xx:56510;transport=tls
Call-ID: 1f47994f9d1e4cf2ac8d3f34c6e71413
CSeq: 28504 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink 0.2.10 (Windows)
Content-Type: application/sdp
Content-Length: 432

v=0
o=- 3575612716 3575612716 IN IP4 192.168.xx.xx
s=Blink 0.2.10 (Windows)
c=IN IP4 192.168.xx.xx
t=0 0
m=audio 50000 RTP/SAVP 0 8 101
a=rtcp:50001
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HYMiBKXZ5Y+t7Fp2H/dXWsNK/zNB299Z27TiqK+h
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:Qpk+q7cnqIbmTOpTGh+6EJDfL9rsI9eSpuBk3rXr
a=sendrecv
<------------->
— (13 headers 14 lines) —
Sending to 84.111.xx.xx:56511 (no NAT)
Using INVITE request as basis request - 1f47994f9d1e4cf2ac8d3f34c6e71413
Found peer ‘20000’ for ‘20000’ from 84.111.xx.xx:56511

<— Reliably Transmitting (NAT) to 84.111.xx.xx:56511 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/tls 192.168.xx.xx:56511;branch=z9hG4bKPj1b3f7663735d448a9e6c9f153ccffe67;received=84.111.xx.xx;rport=56511
From: “20000” sip:20000@173.203.xx.xx;tag=ac06ae02257c4b20b20dda73e286dc4e
To: sip:*97@173.203.xx.xx;tag=as370571e3
Call-ID: 1f47994f9d1e4cf2ac8d3f34c6e71413
CSeq: 28504 INVITE
Server: FPBX2.11.0beta3(11.2.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6ee07b7d"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1f47994f9d1e4cf2ac8d3f34c6e71413’ in 17152 ms (Method: INVITE)

<— SIP read from TLS:84.111.xx.xx:56511 —>
ACK sip:*97@173.203.xx.xx SIP/2.0
Via: SIP/2.0/tls 192.168.xx.xx:56511;rport;branch=z9hG4bKPj1b3f7663735d448a9e6c9f153ccffe67
Max-Forwards: 70
From: “20000” sip:20000@173.203.xx.xx;tag=ac06ae02257c4b20b20dda73e286dc4e
To: sip:*97@173.203.xx.xx;tag=as370571e3
Call-ID: 1f47994f9d1e4cf2ac8d3f34c6e71413
CSeq: 28504 ACK
User-Agent: Blink 0.2.10 (Windows)
Content-Length: 0

<------------->
— (9 headers 0 lines) —

<— SIP read from TLS:84.111.xx.xx:56511 —>
INVITE sip:*97@173.203.xx.xx SIP/2.0
Via: SIP/2.0/tls 192.168.xx.xx:56511;rport;branch=z9hG4bKPj3d749303a42c40d4aff5690cac8735cc
Max-Forwards: 70
From: “20000” sip:20000@173.203.xx.xx;tag=ac06ae02257c4b20b20dda73e286dc4e
To: sip:*97@173.203.xx.xx
Contact: sip:aeljrvms@192.168.xx.xx:56510;transport=tls
Call-ID: 1f47994f9d1e4cf2ac8d3f34c6e71413
CSeq: 28505 INVITE
Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER
Supported: 100rel, replaces, norefersub, gruu
User-Agent: Blink 0.2.10 (Windows)
Authorization: Digest username=“20000”, realm=“asterisk”, nonce=“6ee07b7d”, uri=“sip:*97@173.203.xx.xx”, response=“e6943ea26e129569e75c7d06e8ad044f”, algorithm=MD5
Content-Type: application/sdp
Content-Length: 432

v=0
o=- 3575612716 3575612716 IN IP4 192.168.xx.xx
s=Blink 0.2.10 (Windows)
c=IN IP4 192.168.xx.xx
t=0 0
m=audio 50000 RTP/SAVP 0 8 101
a=rtcp:50001
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:HYMiBKXZ5Y+t7Fp2H/dXWsNK/zNB299Z27TiqK+h
a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:Qpk+q7cnqIbmTOpTGh+6EJDfL9rsI9eSpuBk3rXr
a=sendrecv
<------------->
— (14 headers 14 lines) —
Sending to 84.111.xx.xx:56511 (NAT)
Using INVITE request as basis request - 1f47994f9d1e4cf2ac8d3f34c6e71413
Found peer ‘20000’ for ‘20000’ from 84.111.xx.xx:56511
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
[2013-04-22 06:45:32] WARNING[2936][C-000003bf]: sip/sdp_crypto.c:170 sdp_crypto_activate: Could not set SRTP policies
[2013-04-22 06:45:32] WARNING[2936][C-000003bf]: sip/sdp_crypto.c:170 sdp_crypto_activate: Could not set SRTP policies
[2013-04-22 06:45:32] WARNING[2936][C-000003bf]: chan_sip.c:10427 process_sdp: Can’t provide secure audio requested in SDP offer

<— Reliably Transmitting (NAT) to 84.111.28.144:56511 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/tls 192.168.xx.xx:56511;branch=z9hG4bKPj3d749303a42c40d4aff5690cac8735cc;received=84.111.xx.xx;rport=56511
From: “20000” sip:20000@173.203.xx.xx;tag=ac06ae02257c4b20b20dda73e286dc4e
To: sip:*97@173.203.xx.xx;tag=as370571e3
Call-ID: 1f47994f9d1e4cf2ac8d3f34c6e71413
CSeq: 28505 INVITE
Server: FPBX2.11.0beta3(11.2.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1f47994f9d1e4cf2ac8d3f34c6e71413’ in 17152 ms (Method: INVITE)

<— SIP read from TLS:84.111.xx.xx:56511 —>
ACK sip:*97@173.203.xx.xx SIP/2.0
Via: SIP/2.0/tls 192.168.xx.xx:56511;rport;branch=z9hG4bKPj3d749303a42c40d4aff5690cac8735cc
Max-Forwards: 70
From: “20000” sip:20000@173.203.xx.xx2;tag=ac06ae02257c4b20b20dda73e286dc4e
To: sip:*97@173.203.80.182;tag=as370571e3
Call-ID: 1f47994f9d1e4cf2ac8d3f34c6e71413
CSeq: 28505 ACK
User-Agent: Blink 0.2.10 (Windows)
Content-Length: 0

<------------->
— (9 headers 0 lines) —[/code]

I am really appreciate your assistance.

Thanks,
Noy

[david55]

Is the module, res_srtp, loaded?

[noy_siam]

Yes, its loaded.

SRV*CLI> module show Module Description Use Count app_adsiprog.so Asterisk ADSI Programming Application 0 app_alarmreceiver.so Alarm Receiver for Asterisk 0 app_amd.so Answering Machine Detection Application 0 app_authenticate.so Authentication Application 0 app_cdr.so Tell Asterisk to not maintain a CDR for 0 app_celgenuserevent.so Generate an User-Defined CEL event 0 app_chanisavail.so Check channel availability 0 app_channelredirect.so Redirects a given channel to a dialplan 0 app_chanspy.so Listen to the audio of an active channel 0 app_confbridge.so Conference Bridge Application 0 app_controlplayback.so Control Playback Application 0 app_dahdiras.so DAHDI ISDN Remote Access Server 0 app_db.so Database Access Functions 0 app_dial.so Dialing Application 0 app_dictate.so Virtual Dictation Machine 0 app_directed_pickup.so Directed Call Pickup Application 0 app_directory.so Extension Directory 0 app_disa.so DISA (Direct Inward System Access) Appli 0 app_dumpchan.so Dump Info About The Calling Channel 0 app_echo.so Simple Echo Application 0 app_exec.so Executes dialplan applications 0 app_externalivr.so External IVR Interface Application 0 app_festival.so Simple Festival Interface 0 app_flash.so Flash channel application 0 app_followme.so Find-Me/Follow-Me Application 0 app_forkcdr.so Fork The CDR into 2 separate entities 0 app_getcpeid.so Get ADSI CPE ID 0 app_ices.so Encode and Stream via icecast and ices 0 app_image.so Image Transmission Application 0 app_macro.so Extension Macros 0 app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 app_minivm.so Mini VoiceMail (A minimal Voicemail e-ma 0 app_mixmonitor.so Mixed Audio Monitoring Application 0 app_morsecode.so Morse code 0 app_mp3.so Silly MP3 Application 0 app_nbscat.so Silly NBS Stream Application 0 app_originate.so Originate call 0 app_page.so Page Multiple Phones 0 app_parkandannounce.so Call Parking and Announce Application 0 app_playback.so Sound File Playback Application 0 app_playtones.so Playtones Application 0 app_privacy.so Require phone number to be entered, if n 0 app_queue.so True Call Queueing 0 app_read.so Read Variable Application 0 app_readexten.so Read and evaluate extension validity 0 app_record.so Trivial Record Application 0 app_sayunixtime.so Say time 0 app_senddtmf.so Send DTMF digits Application 0 app_sendtext.so Send Text Applications 0 app_sms.so SMS/PSTN handler 0 app_softhangup.so Hangs up the requested channel 0 app_speech_utils.so Dialplan Speech Applications 0 app_stack.so Dialplan subroutines (Gosub, Return, etc 0 app_system.so Generic System() application 0 app_talkdetect.so Playback with Talk Detection 0 app_test.so Interface Test Application 0 app_transfer.so Transfers a caller to another extension 0 app_url.so Send URL Applications 0 app_userevent.so Custom User Event Application 0 app_verbose.so Send verbose output 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_waitforring.so Waits until first ring after time 0 app_waitforsilence.so Wait For Silence 0 app_waituntil.so Wait until specified time 0 app_while.so While Loops and Conditional Execution 0 app_zapateller.so Block Telemarketers with Special Informa 0 bridge_builtin_features.so Built in bridging features 1 bridge_multiplexed.so Multiplexed two channel bridging module 0 bridge_simple.so Simple two channel bridging module 0 bridge_softmix.so Multi-party software based channel mixin 0 cdr_csv.so Comma Separated Values CDR Backend 0 cdr_custom.so Customizable Comma Separated Values CDR 0 cdr_manager.so Asterisk Manager Interface CDR Backend 0 cdr_sqlite3_custom.so SQLite3 Custom CDR Module 0 cdr_syslog.so Customizable syslog CDR Backend 0 cel_custom.so Customizable Comma Separated Values CEL 0 cel_manager.so Asterisk Manager Interface CEL Backend 0 cel_sqlite3_custom.so SQLite3 Custom CEL Module 0 chan_agent.so Agent Proxy Channel 0 chan_bridge.so Bridge Interaction Channel 0 chan_dahdi.so DAHDI Telephony Driver w/PRI 0 chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 chan_local.so Local Proxy Channel (Note: used internal 0 chan_multicast_rtp.so Multicast RTP Paging Channel 0 chan_oss.so OSS Console Channel Driver 0 chan_phone.so Linux Telephony API Support 0 chan_sip.so Session Initiation Protocol (SIP) 0 chan_skinny.so Skinny Client Control Protocol (Skinny) 0 chan_unistim.so UNISTIM Protocol (USTM) 0 codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 codec_alaw.so A-law Coder/Decoder 0 codec_dahdi.so Generic DAHDI Transcoder Codec Translato 0 codec_g722.so ITU G.722-64kbps G722 Transcoder 0 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 codec_gsm.so GSM Coder/Decoder 0 codec_ilbc.so iLBC Coder/Decoder 0 codec_lpc10.so LPC10 2.4kbps Coder/Decoder 0 codec_resample.so SLIN Resampling Codec 0 codec_ulaw.so mu-Law Coder/Decoder 0 format_g719.so ITU G.719 0 format_g723.so G.723.1 Simple Timestamp File Format 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 format_g729.so Raw G.729 data 0 format_gsm.so Raw GSM data 0 format_h263.so Raw H.263 data 0 format_h264.so Raw H.264 data 0 format_ilbc.so Raw iLBC data 0 format_jpeg.so jpeg (joint picture experts group) image 0 format_pcm.so Raw/Sun uLaw/ALaw 8KHz (PCM,PCMA,AU), G. 0 format_siren14.so ITU G.722.1 Annex C (Siren14, licensed f 0 format_siren7.so ITU G.722.1 (Siren7, licensed from Polyc 0 format_sln.so Raw Signed Linear Audio support (SLN) 8k 0 format_vox.so Dialogic VOX (ADPCM) File Format 0 format_wav.so Microsoft WAV/WAV16 format (8kHz/16kHz S 0 format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 func_aes.so AES dialplan functions 0 func_audiohookinherit.so Audiohook inheritance function 0 func_base64.so base64 encode/decode dialplan functions 0 func_blacklist.so Look up Caller*ID name/number from black 0 func_callcompletion.so Call Control Configuration Function 0 func_callerid.so Party ID related dialplan functions (Cal 0 func_cdr.so Call Detail Record (CDR) dialplan functi 0 func_channel.so Channel information dialplan functions 0 func_config.so Asterisk configuration file variable acc 0 func_cut.so Cut out information from a string 0 func_db.so Database (astdb) related dialplan functi 0 func_devstate.so Gets or sets a device state in the dialp 0 func_dialgroup.so Dialgroup dialplan function 0 func_dialplan.so Dialplan Context/Extension/Priority Chec 0 func_enum.so ENUM related dialplan functions 0 func_env.so Environment/filesystem dialplan function 0 func_extstate.so Gets an extension's state in the dialpla 0 func_frame_trace.so Frame Trace for internal ast_frame debug 0 func_global.so Variable dialplan functions 0 func_groupcount.so Channel group dialplan functions 0 func_hangupcause.so HANGUPCAUSE related functions and applic 0 func_iconv.so Charset conversions 0 func_jitterbuffer.so Jitter buffer for read side of channel. 0 func_lock.so Dialplan mutexes 0 func_logic.so Logical dialplan functions 0 func_math.so Mathematical dialplan function 0 func_md5.so MD5 digest dialplan functions 0 func_module.so Checks if Asterisk module is loaded in m 0 func_pitchshift.so Audio Effects Dialplan Functions 0 func_presencestate.so Gets or sets a presence state in the dia 0 func_rand.so Random number dialplan function 0 func_realtime.so Read/Write/Store/Destroy values from a R 0 func_sha1.so SHA-1 computation dialplan function 0 func_shell.so Collects the output generated by a comma 0 func_sprintf.so SPRINTF dialplan function 0 func_srv.so SRV related dialplan functions 0 func_strings.so String handling dialplan functions 0 func_sysinfo.so System information related functions 0 func_timeout.so Channel timeout dialplan functions 0 func_uri.so URI encode/decode dialplan functions 0 func_version.so Get Asterisk Version/Build Info 0 func_vmcount.so Indicator for whether a voice mailbox ha 0 func_volume.so Technology independent volume control 0 pbx_ael.so Asterisk Extension Language Compiler 0 pbx_config.so Text Extension Configuration 0 pbx_dundi.so Distributed Universal Number Discovery ( 0 pbx_loopback.so Loopback Switch 0 pbx_realtime.so Realtime Switch 0 pbx_spool.so Outgoing Spool Support 0 res_adsi.so ADSI Resource 0 res_ael_share.so share-able code for AEL 0 res_agi.so Asterisk Gateway Interface (AGI) 1 res_calendar.so Asterisk Calendar integration 0 res_clialiases.so CLI Aliases 0 res_clioriginate.so Call origination and redirection from th 0 res_config_sqlite3.so SQLite 3 realtime config engine 0 res_convert.so File format conversion CLI command 0 res_crypto.so Cryptographic Digital Signatures 0 res_fax.so Generic FAX Applications 0 res_format_attr_celt.so CELT Format Attribute Module 0 res_format_attr_h263.so H.263 Format Attribute Module 0 res_format_attr_h264.so H.264 Format Attribute Module 0 res_format_attr_silk.so SILK Format Attribute Module 0 res_http_websocket.so HTTP WebSocket Support 0 res_limit.so Resource limits 0 res_monitor.so Call Monitoring Resource 0 res_musiconhold.so Music On Hold Resource 0 res_mutestream.so Mute audio stream resources 0 res_phoneprov.so HTTP Phone Provisioning 0 res_realtime.so Realtime Data Lookup/Rewrite 0 res_rtp_asterisk.so Asterisk RTP Stack 0 res_rtp_multicast.so Multicast RTP Engine 0 res_security_log.so Security Event Logging 0 res_smdi.so Simplified Message Desk Interface (SMDI) 0 res_speech.so Generic Speech Recognition API 0 res_srtp.so Secure RTP (SRTP) 0 res_stun_monitor.so STUN Network Monitor 0 res_timing_dahdi.so DAHDI Timing Interface 0 res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 196 modules loaded

[david55]

That module appears to have rejected the SRTP setup, but there doesn’t seem to be much debugging available from it, so working out what is wrong will require more code reading time than I’m prepared to give.

[noy_siam]

Thank you very much for your time David.
I have two question about it, if you don’t mind.
Do you think should I install all again from scratch?
Do you have or know about tutorial to do it?
I just need the SRTP working and I do not care which asterisk version I am installing.

Regards,
Noy

[system]

Well it looks like you have forgotten to install the patch which is present at the same place you downloaded the source from.

untar and unzip your asterisk source

patch it with its corresponding version

./configure

make clean

make menuselect

make

make install

amportal restart

And yes you have to compile it again like above

I had a similar issue that I resolved patching my asterisk installation.

[navaismo]

Which patch? Asterisk don’t need a patch to work with tls. Also notice that amportal is a command for FreePBX based distribution not for asterisk .

[java_guy]

I have pathced

patch -p0 -i ./asterisk_379070.patch

Should i remove asterisk from system and install it without the patch?

[mixailom]

Did you manage to have this working?

I have the same problem WARNING[22359][C-00000082]: chan_sip.c:10648 process_sdp: Can’t provide secure audio requested in SDP offer