[Spce-user] webRTC in production

[Nikita Stashkov]

You need to look logs from WebRTC client and Pcap from SIP.
Of cource, if SIP client recives SDP with rtp-mux, he will not understand it. And after 30 sec call will be terminated. But you must see logs. My system is based on SPCE 3.2, and manually compiled rtpengine. And I don't know was changed in current version. Also, you can look rtp.log. Sometimes it helps.

Regards,
Nikita Stashkov


> 23. nov. 2014, в 13.40, H Yavari <hyavari at rocketmail.com> написал(а):
> 
> Hi,
> 
> Dear I did this before that I changed "ws" with "wss" but now after your reply I did "ws" || "wss". but not any changes.
> As I told before, now my main problem is calls hangup after 30 sec. In your opinion the rtcp-mux-demux flags adding will solve this?
> another point is that before 30 sec, If any call parties (caller: browser and callee: soft phone) hangs up, the call not terminate until 30 sec timeout. I think that the dialog of a call not recognized.
> 
> So situation is complicated :)
> SPCE specialist plz help!
> 
> 
> Regards,
> H. Yavari
> 
> From: Nikita Stashkov <snl at sipmobile.org>
> 
> Sorry, I can not share my script.
> What can you do.
> Look the script /etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.tt2
> Of course, before modifying copy it to proxy.cfg.customtt.tt2
> I think, webrtc endpoint automatic detection is not working for you.
> It must look like this:
> 
> if($(ru{uri.param,transport}) == "ws" || $(ru{uri.param,transport}) == "wss»)
> 
> Then check flags you are sending to rtpengine.
> To call SIP clients you must use flag rtcp-mux-demux
> 
> Regards,
> Nikita Stashkov
> 
> 
>> 22 нояб. 2014 г., в 20:28, H Yavari <hyavari at rocketmail.com> написал(а):
>> 
>> 
>> 
>> Hi,
>> I checked you site. it seems that is a good webRTC solution.
>> Can you share with us your experience to solve our problem? or any script modifications?
>> 
>> 
>> Regards,
>> H.Yavari
>> 
>> From: Nikita Stashkov <snl at sipmobile.org>
>> 
>> You can try with my site - www.sipmobile.org.
>> Create accounts: 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>:
>>> 
>>> 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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>> 
>> 
>> 
> 
> 
> 
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