[Spce-user] webRTC in production

Mon Nov 24 06:09:00 EST 2014

And the first one is sipml5?
In my config both are working.
Check again your script. There is not one place, where automatic detection is done.
I don’t exactly remember. It was about 4-5 month ago. But I think, difference is between ws and wss.
Sorry, I can not publish my script. There are many other things, including push notifications.
I think Sipwise will be not happy, if I publish it.

Regards,
Nikita Stashkov
> 24 нояб. 2014 г., в 11:45, H Yavari <hyavari at rocketmail.com> написал(а):
> 
> Hi,
> 
> I noticed a new thing that when I using jssip, calls not terminated. in the logs and I didn't see any rtcp-mux. so this two webRTC client is different in using SDP params?
> 
> 
> Regards,
> H.Yavari
> From: Nikita Stashkov <snl at sipmobile.org>
> 
> 
> There may be different logic. I am doing it if caller is WebRTC, and callee is SIP. 
> Simply add this flag, when calling Rtpengine, like all other flags.
> You can do nothing if both are WebRTC.
> 
> Regards,
> Nikita Stashkov
> 
> 
> 
> 24 нояб. 2014 г., в 11:13, H Yavari <hyavari at rocketmail.com <mailto:hyavari at rocketmail.com>> написал(а):
> 
> 
> 
>> Hi,
>> I checked it. the client (webRTC browser-sipml5) send rtp-mux. and is in the rtp.log too. so how can I disable this? or how can I add rtcp-mux-demux ? I should do this for all calls? or only for webRTC client?
>> 
>> 
>> Thanks a lot.
>> 
>> Regards,
>> H.Yavari
>> 
>> From: Nikita Stashkov <snl at sipmobile.org <mailto:snl at sipmobile.org>>
>> 
>> 
>> 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 <mailto: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 <mailto: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 <mailto: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 <mailto:snl at sipmobile.org>>
>>>> 
>>>> 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 <mailto: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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>>>> 
>>>> 
>>>> 
>>> 
>>> 
>>> 
>> 
>> 
> 
> 

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