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

Victor Medina vittico at gmail.com
Fri May 8 01:17:22 MSD 2015


Hi Giovanni.

Thanks for the replay.

You mean this confluence page?
https://freeswitch.org/confluence/display/FREESWITCH/WebRTC


Sin mas a que hacer referencia,

Victor Medina

On Thu, May 7, 2015 at 4:35 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

> 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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>>
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>>
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>>
>
> _________________________________________________________________________
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>
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