[Freeswitch-users] Testing WebRTC with latest master Git 1.5

Fri May 8 01:05:07 MSD 2015

Start by scratch, from a fresh debian Jessie install.

Then follow exactly, without changes, step by step, what is in the
"freeswitch 1.6" confluence page.

When you have it all working as in that page (contains instruction on
testing) only then you can go further.

Happy testing,

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 7, 2015 10:58 PM, "Victor Medina" <victor.medina at cibersys.com> wrote:

> Hi guys!
>
> I´ve been tying to test Freeswitch 1.5 (latest git) for WebRTC, but so far
> I have been very unsuccessful.
>
> I have a server, connected directly to internet, NO NAT on server side.
>
>
> FS ----> INTERNET <--- NAT ---- CLIENTS
>
> My vars.conf includes this:
>
>
>   <X-PRE-PROCESS cmd="set" data="bind_server_ip=xxx.xxx.xxx.xxx"/>
>   <X-PRE-PROCESS cmd="set" data="external_rtp_ip=xxx.xxx.xxx.xxx"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_ip=xxxx.xxx.xxx.xxx"/>
>
>   <!-- Internal SIP Profile -->
>   <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/>
>   <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>
>   <X-PRE-PROCESS cmd="set" data="internal_tls_port=5061"/>
>   <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/>
>   <X-PRE-PROCESS cmd="set"
> data="internal_ssl_dir=/opt/CloudVoice-vPBX/fs-20150506/certs/"/>
>
>   <!-- External SIP Profile -->
>   <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
>   <X-PRE-PROCESS cmd="set" data="external_sip_port=5080"/>
>   <X-PRE-PROCESS cmd="set" data="external_tls_port=5081"/>
>   <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/>
>
>
> On my external profile I have this relevants lines...
>
>  <param name="aggressive-nat-detection" value="true"/>
>  <param name="apply-inbound-acl" value="domains"/>
>  <param name="local-network-acl" value="localnet.auto"/>
>  <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>
>  <param name="ext-sip-ip" value="$${external_sip_ip}"/>
>  <param name="ws-binding"  value=":5066"/>
>  <param name="wss-binding" value=":7443"/>
>
>
> When doing some testing....
>
> Calling to echo test
>
>  2015-05-07 20:51:14.124955 [NOTICE] switch_channel.c:1075 New Channel
> sofia/internal/1007 at webrtc.cibersys.com
> [39b777d1-6e61-49ec-bfd0-6f7ca0bd0caf]
> 2015-05-07 20:51:14.384990 [INFO] mod_dialplan_xml.c:635 Processing test
> <1007>->9196 in context default
> 2015-05-07 20:51:14.384990 [WARNING] switch_core_media.c:2791 NO candidate
> ACL defined, Defaulting to wan.auto
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2953 setting
> remote audio ice addr to 201.210.31.83:57630 based on candidate
> 2015-05-07 20:51:14.384990 [NOTICE] switch_core_media.c:2978 setting
> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5345 Activating
> Audio ICE
> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTP audio
> ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5388 Activating RTCP
> PORT 57632
> 2015-05-07 20:51:14.384990 [INFO] switch_core_media.c:5398 Activating RTCP
> ICE
> 2015-05-07 20:51:14.384990 [NOTICE] switch_rtp.c:4019 Activating RTCP
> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
> 2015-05-07 20:51:14.384990 [INFO] switch_rtp.c:3103 Activate RTP/RTCP
> audio DTLS client
> 2015-05-07 20:51:14.384990 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/internal/1007 at webrtc.cibersys.com!
> 2015-05-07 20:51:14.384990 [NOTICE] mod_dptools.c:1292 Channel
> [sofia/internal/1007 at webrtc.cibersys.com] has been answered
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2924 Changing audio DTLS
> state from HANDSHAKE to SETUP
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2832 audio Fingerprint
> Verified.
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3384 Activating Audio
> Secure RTP SEND
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:3362 Activating Audio
> Secure RTP RECV
> 2015-05-07 20:51:14.864993 [INFO] switch_rtp.c:2872 Changing audio DTLS
> state from SETUP to READY
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
> audio RTCP PORT 57632
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: UDP type: host addr: 10.0.0.126:57630
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 1 proto: UDP type: host addr: 192.168.56.1:57631
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: UDP type: host addr: 10.0.0.126:57632
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2829 Save audio
> Candidate cid: 2 proto: UDP type: host addr: 192.168.56.1:57633
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 1 proto: UDP type: srflx addr: 201.210.31.83:57630
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2824 Choose audio
> Candidate cid: 2 proto: UDP type: srflx addr: 201.210.31.83:57632
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2953 setting
> remote audio ice addr to 201.210.31.83:57630 based on candidate
> 2015-05-07 20:52:15.564950 [NOTICE] switch_core_media.c:2978 setting
> remote rtcp audio addr to 201.210.31.83:57632 based on candidate
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:2995 RE-Activating
> audio ICE
> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTP audio
> ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57630
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3039 Activating
> audio RTCP PORT 57632
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:3050 Activating
> audio RTCP ICE
> 2015-05-07 20:52:15.564950 [NOTICE] switch_rtp.c:4019 Activating RTCP
> audio ICE: 76ffb71e:TduhdjNUFKq3Vzaz 201.210.31.83:57632
> 2015-05-07 20:52:15.564950 [INFO] switch_core_media.c:5030 RE-SETTING
> video DTLS
> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:878 sofia/internal/
> 1007 at webrtc.cibersys.com got stun binding response 487 Role Conflict
> 2015-05-07 20:52:15.704955 [WARNING] switch_rtp.c:889 Changing role to
> CONTROLLED
>
> No audio
>
> When calling another ext... no audio en the webrtc side.
>
> Can somebody help me by pointing out the right direction?
>
> Ive been using FF and Chrome with sipML5
>
>
>
> --
>
>
>
> Víctor E. Medina M.
> Platform Architect / Chief Infrastructure
> +58424 291 4561
> BB #79A8AFA2
> @VMCibersys
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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