Hi everyone I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013') I'm using the default configuration and just uncommentated the ' <param name="ws-binding" value=":5066"/> ' in internal.xml in order to have the support for webrtc. As the client I'm having JSSIP, the latest version with the adjustment to have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound: [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client: 2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/ 1000@10.0.14.16:5060 timer while HOT 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/ 1000@10.0.14.16:5060 Hot Hit 1 And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts: 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/ sip:1000@10.0.14.182:5065 Hot Hit 4 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/ sip:1000@10.0.14.182:5065 timer while HOT 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!) If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000! Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? Thanks in advanced! Sherry
We only use non rfc-1918 ip's by default. If you want to use 1918 ip's you need to tweak acls Mike On Aug 5, 2013, at 8:24 AM, Shahrzad A. <shahrzad.aziminia@gmail.com> wrote:
Hi everyone
I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013') I'm using the default configuration and just uncommentated the ' <param name="ws-binding" value=":5066"/> ' in internal.xml in order to have the support for webrtc. As the client I'm having JSSIP, the latest version with the adjustment to have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound:
[ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)
its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client:
2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/1000@10.0.14.16:5060 timer while HOT 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/1000@10.0.14.16:5060 Hot Hit 1
And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts:
2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/sip:1000@10.0.14.182:5065 Hot Hit 4 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/sip:1000@10.0.14.182:5065 timer while HOT 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)
If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000!
Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP?
Thanks in advanced!
Sherry
Hi Michael Thanks for your reply, Would you please advice me how to change the ACL to fix this issue. Here is my log when I try to call from a JSSIP/Sipml5 to another JSSIP/Sipml5 client: 2013-08-27 13:33:37.742425 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/1008@10.0.14.16 [771e9dfb-ef68-4adc-9e5a-380e172e1945] 2013-08-27 13:33:37.762422 [INFO] mod_dialplan_xml.c:558 Processing Lala <1008>->1004 in context public 2013-08-27 13:33:37.762422 [NOTICE] switch_ivr.c:1831 Transfer sofia/internal/1008@10.0.14.16 to XML[1004@default] 2013-08-27 13:33:37.762422 [INFO] mod_dialplan_xml.c:558 Processing Lala <1008>->1004 in context default 2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1 execute_extension::dx XML features 2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2 record_session::/home/saz/freeService/recordings/1008.2013-08-27-13-33-37.wav 2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3 execute_extension::cf XML features 2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4 execute_extension::att_xfer XML features 2013-08-27 13:33:37.762422 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/sip:1004@df7jal23ls0d.invalid [876787a6-a8f5-4d1c-bd98-ab2246a9114f] 2013-08-27 13:33:37.802411 [NOTICE] sofia.c:5898 Ring-Ready sofia/internal/sip:1004@df7jal23ls0d.invalid! 2013-08-27 13:33:37.802411 [WARNING] switch_channel.c:3250 rtp_secure_media invalid in this context. 2013-08-27 13:33:37.802411 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-08-27 13:33:37.802411 [WARNING] switch_core_media.c:2074 NO candidate ACL defined, Defaulting to wan.auto 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:33865 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 2 proto: udp type: host addr: 10.0.14.254:33865 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:53815 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 2 proto: udp type: host addr: 10.0.14.253:53815 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2221 setting remote audio ice addr to 10.0.14.254:33865 based on candidate 2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2239 setting remote rtcp audio addr to 10.0.14.254:33865 based on candidate 2013-08-27 13:33:37.822462 [INFO] switch_core_media.c:4227 Activating Audio ICE 2013-08-27 13:33:37.822462 [NOTICE] switch_rtp.c:3323 Activating RTP audio ICE: ruUP7F49vDADu+Gp:GMkGzis5aAzjoC0v 10.0.14.254:33865 2013-08-27 13:33:37.822462 [INFO] switch_core_media.c:4270 Activating RTCP PORT 33865 2013-08-27 13:33:37.822462 [INFO] switch_core_media.c:4278 Skipping RTCP ICE (Same as RTP) 2013-08-27 13:33:37.822462 [INFO] switch_rtp.c:2711 Activating Audio Secure RTP SEND 2013-08-27 13:33:37.822462 [INFO] switch_rtp.c:2689 Activating Audio Secure RTP RECV 2013-08-27 13:33:37.822462 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1008@10.0.14.16! 2013-08-27 13:33:37.922423 [NOTICE] switch_rtp.c:1053 Auto Changing stun/rtp/dtls port from 10.0.14.254:33865 to 10.0.14.253:53815 2013-08-27 13:33:42.342422 [WARNING] switch_core_media.c:2074 NO candidate ACL defined, Defaulting to wan.auto 2013-08-27 13:33:42.342422 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:45217 2013-08-27 13:33:42.342422 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:38004 2013-08-27 13:33:42.342422 [NOTICE] switch_core_media.c:2221 setting remote audio ice addr to 10.0.14.254:45217 based on candidate 2013-08-27 13:33:42.342422 [INFO] switch_core_media.c:4227 Activating Audio ICE 2013-08-27 13:33:42.342422 [NOTICE] switch_rtp.c:3323 Activating RTP audio ICE: gd9dRK830iEDLw2W:vECHRV23TjEY0c0I 10.0.14.254:45217 2013-08-27 13:33:42.342422 [INFO] switch_core_media.c:4270 Activating RTCP PORT 45217 2013-08-27 13:33:42.342422 [INFO] switch_rtp.c:2478 Activate RTP/RTCP audio DTLS server 2013-08-27 13:33:42.342422 [NOTICE] sofia.c:6561 Channel [sofia/internal/sip:1004@df7jal23ls0d.invalid] has been answered 2013-08-27 13:33:42.362407 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1008@10.0.14.16] has been answered As you can see there is no DTLS or handshake! But the interesting part is, when I try to call from SIPML5 client to a JSSIP client I have the following log with handshake and the voice is perfect! would you please tell me what i'm missing! 2013-08-27 13:37:05.242424 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/1008@10.0.14.16 [14f54a7b-6077-4bc3-b845-0b8bbfa3d59d] 2013-08-27 13:37:05.282411 [INFO] mod_dialplan_xml.c:558 Processing Lala <1008>->1003 in context public 2013-08-27 13:37:05.282411 [NOTICE] switch_ivr.c:1831 Transfer sofia/internal/1008@10.0.14.16 to XML[1003@default] 2013-08-27 13:37:05.282411 [INFO] mod_dialplan_xml.c:558 Processing Lala <1008>->1003 in context default 2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1 execute_extension::dx XML features 2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2 record_session::/home/saz/freeService/recordings/1008.2013-08-27-13-37-05.wav 2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3 execute_extension::cf XML features 2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4 execute_extension::att_xfer XML features 2013-08-27 13:37:05.282411 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/sip:75icg9r5@i3ifqjalvn7j.invalid [d1fb6893-6870-4b5d-8d2c-28b038df9965] 2013-08-27 13:37:05.322412 [NOTICE] sofia.c:5898 Ring-Ready sofia/internal/sip:75icg9r5@i3ifqjalvn7j.invalid! 2013-08-27 13:37:05.322412 [WARNING] switch_channel.c:3250 rtp_secure_media invalid in this context. 2013-08-27 13:37:05.322412 [INFO] switch_ivr_originate.c:1190 Sending early media 2013-08-27 13:37:05.322412 [WARNING] switch_core_media.c:2074 NO candidate ACL defined, Defaulting to wan.auto 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:49236 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 2 proto: udp type: host addr: 10.0.14.254:49236 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:40898 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 2 proto: udp type: host addr: 10.0.14.253:40898 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2221 setting remote audio ice addr to 10.0.14.254:49236 based on candidate 2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2239 setting remote rtcp audio addr to 10.0.14.254:49236 based on candidate 2013-08-27 13:37:05.322412 [INFO] switch_core_media.c:4227 Activating Audio ICE 2013-08-27 13:37:05.322412 [NOTICE] switch_rtp.c:3323 Activating RTP audio ICE: 9nidt62aGODady6Y:Hbj7Rd3CDzM2yOIn 10.0.14.254:49236 2013-08-27 13:37:05.322412 [INFO] switch_core_media.c:4270 Activating RTCP PORT 49236 2013-08-27 13:37:05.322412 [INFO] switch_core_media.c:4278 Skipping RTCP ICE (Same as RTP) 2013-08-27 13:37:05.322412 [INFO] switch_rtp.c:2711 Activating Audio Secure RTP SEND 2013-08-27 13:37:05.322412 [INFO] switch_rtp.c:2689 Activating Audio Secure RTP RECV 2013-08-27 13:37:05.322412 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1008@10.0.14.16! 2013-08-27 13:37:17.302425 [WARNING] switch_core_media.c:2074 NO candidate ACL defined, Defaulting to wan.auto 2013-08-27 13:37:17.302425 [NOTICE] switch_core_media.c:2107 Choose audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:48285 2013-08-27 13:37:17.302425 [NOTICE] switch_core_media.c:2112 Save audio Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:53583 2013-08-27 13:37:17.302425 [NOTICE] switch_core_media.c:2221 setting remote audio ice addr to 10.0.14.254:48285 based on candidate 2013-08-27 13:37:17.302425 [INFO] switch_core_media.c:4227 Activating Audio ICE 2013-08-27 13:37:17.302425 [NOTICE] switch_rtp.c:3323 Activating RTP audio ICE: O8IjxMVvzmi10qL4:brG3ubnHrh04oRt9 10.0.14.254:48285 2013-08-27 13:37:17.302425 [INFO] switch_core_media.c:4270 Activating RTCP PORT 48285 2013-08-27 13:37:17.302425 [INFO] switch_rtp.c:2478 Activate RTP/RTCP audio DTLS server 2013-08-27 13:37:17.302425 [NOTICE] sofia.c:6561 Channel [sofia/internal/sip:75icg9r5@i3ifqjalvn7j.invalid] has been answered 2013-08-27 13:37:17.322410 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1008@10.0.14.16] has been answered 2013-08-27 13:37:19.322423 [NOTICE] switch_rtp.c:1053 Auto Changing stun/rtp/dtls port from 10.0.14.254:48285 to 10.0.14.253:53583 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2370 Changing audio DTLS state from HANDSHAKE to SETUP 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2289 audio Fingerprint Verified. 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2711 Activating Audio Secure RTP SEND 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2689 Activating Audio Secure RTP RECV 2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2329 Changing audio DTLS state from SETUP to READY 2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2391 Changing audio DTLS state from READY to HANDSHAKE 2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2370 Changing audio DTLS state from HANDSHAKE to SETUP 2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2289 audio Fingerprint Verified. 2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2329 Changing audio DTLS state from SETUP to READY -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Issue-with-JSSIP-Freeswitch-tp... Sent from the freeswitch-users mailing list archive at Nabble.com.
participants (3)
-
Michael Jerris
-
Shahrzad A.
-
sherrylist