[Spce-user] webRTC in production

[Nikita Stashkov]

You can try with my site - www.sipmobile.org <http://www.sipmobile.org/>.
Create accounts: https://www.sipmobile.org/register/ <https://www.sipmobile.org/register/>
And try to call with webRTC client and SIP.
I have modified some Kamailio SPCE scripts.

Regards,
Nikita Stashkov


> 22 нояб. 2014 г., в 16:04, Thomas Odorfer <odotom at gmail.com> написал(а):
> 
> Hi,
> not sure if I understood correctly which scenario works and which not. 
> So browser to soft phone is now working, but what is the meaning of browser to client? Which client?
> 
> I tested myself and I have to confess that I had to do some changes in the account configs for soft phones where I am not happy about.
> It only worked between browser-webrtc  and soft phone when the corresponding account for the soft phone - nat & media flow control had been changed to "force avp"“ and "force rtp“ ie. no encryption.
> (I have to investigate that one - could be related to an upgrade I had performed last week - usually srtp should also work with softphones, within the ftp.log there was „SRTP output wanted but no crypto suite was negotiated“).
> However, after my changes the following tests had been successful:
> browser webrtc  to  softphone (eg. jitsi,  counterpath x-lite - should be software compatible with eyebeam)
> browser webrtc to  another browser webrtc (jssip-0.50)
> browser webrtc to pstn via sip trunking  (standard sip trunk, peer settings for media  force „rtp“, „force rtp“, „always with plain SDP“)
> 
> That is based on the latest SPCE version 3.6.1.
> What does not seem to be achievable at the moment that you can have an account that supports „standard“ and webrtc simultaneously ( at least I haven’t succeeded with such a setup, maybe some sipwise/kamailio/rtpengine  expert knows the trick). And I do not have a solution yet how to share one phone number between two accounts with different profiles.
> The only solution I have at the moment is that I put a webrtc gateway (similar to webrtc2sip  from doubango) in front of SPCE for webrtc clients.
> 
> For your particular problem, maybe you have to check whether your domain settings allow „bypass rtp proxy“ behind the same NAT - assuming you are testing wthin your LAN - this should be set to never.
> 
> Good luck
> Thomas
> 
> 
> Am 22.11.2014 um 12:49 schrieb H Yavari <hyavari at rocketmail.com <mailto:hyavari at rocketmail.com>>:
> 
>> Hi,
>> 
>> I did this configs:
>> use_rtpproxy:   „Always with rtpptoxy as only ICE candidate“ 
>> rtcp_feedback:  „Force AVP“ 
>> srtp_transcoding:    „Force RTP“
>> 
>> now calls between browser to soft phone is ok, but browser to client and browser to browser receive this error "Failed to get local SDP"
>> and calls not be established. Have you any idea about this situation? 
>> Thanks for helps.
>> 
>> Regards,
>> H.Yavari
>> 
> 
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