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

Giovanni Maruzzelli gmaruzz at gmail.com
Fri May 8 02:49:22 MSD 2015 

Go with the one I told you. It has webrrc too, and the docs are completely
updated, and we all are using and testing that one.
Eg: is the path of less resistance ;)
Then you can use only what part you want (webrtc), but at least you sure if
you follow the steps it will work without fiddling.

sent from my mobile,
Giovanni Maruzzelli
cell: +39 347 266 56 18
On May 8, 2015 12:01 AM, "Victor Medina" <vittico at gmail.com> wrote:

> OK, very impressive....
>
> But... I, for the moment ONLY want to have a _basic_ webrtc softhone
> connected to FS. As I undestand this, is for Video Conferencing right?
>
> Is this https://freeswitch.org/confluence/display/FREESWITCH/WebRTC
> already deprecated? Not working anymore?
>
>
> Sin mas a que hacer referencia,
>
> Victor Medina
>
> On Thu, May 7, 2015 at 4:59 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>
>> No.
>> This one:
>> https://freeswitch.org/confluence/plugins/servlet/mobile#content/view/7144556
>>
>> sent from my mobile,
>> Giovanni Maruzzelli
>> cell: +39 347 266 56 18
>> On May 7, 2015 11:18 PM, "Victor Medina" <vittico at gmail.com> wrote:
>>
>>> 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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>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
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>>
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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
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>
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