[Spce-user] webRTC in production

H Yavari hyavari at rocketmail.com
Tue Nov 25 08:42:19 EST 2014 

I don't see force SRTP option,I set it to Prefer SRTP. I did this like your pdf attachment.
Regards,H. Yavari
     From: Nikita Stashkov <snl at sipmobile.org>
 
   
In your domain settings do you force SRTP?I do.
Regards,Nikita Stashkov

25 нояб. 2014 г., в 14:08, H Yavari <hyavari at rocketmail.com> написал(а):


Hi,Thanks. I see.Have you any idea about this error : "SRTP output wanted, but no crypto suite was negotiated" ???Is this related to dtls handshake and fingerprints?I see this too: https://github.com/sipwise/mediaproxy-ng/blob/master/daemon/rtp.c

Regards,H. Yavari
     From: Andreas Granig <agranig at sipwise.com>
Please see https://github.com/sipwise/rtpengine#offer-message for
available options and their possible values.

Andreas

On 11/25/2014 01:35 PM, H Yavari wrote:
> Hi,
> I copied the all flags same as Nikita script.Nothing has changed but in
> the rtp.log there are some lines :
> Nov 24 07:36:55 spce rtpengine[6426]: Unknown flag encountered: 'symmetric'
> Nov 24 07:36:55 spce rtpengine[6426]: Unknown 'rtcp-mux' flag
> encountered: 'demuxSRTP'
> 
> Nov 24 07:37:00 spce rtpengine[6426]:
> [f5008c55-8329-f08e-e024-81d8260b1708 port 30865] SRTCP output wanted,
> but no crypto suite was negotiated
> .
> .
> .
> .
> Nov 24 07:37:32 spce rtpengine[6426]:
> [f5008c55-8329-f08e-e024-81d8260b1708] Scheduling deletion of call
> branch 'R7t3SMDI7STFq4A53a9w' in 30 seconds
> 
> this flags not supported by rtpengine now? (3.6.1)
> how suite crypto will be negotiated?
> 
> 
> Regards,
> H. Yavari
> 
> 
> ------------------------------------------------------------------------
> *From:* Andreas Granig <agranig at sipwise.com>
> 
> 
> You don't need stun/turn with rtpengine, because it puts itself into the
> SDP as ICE candidate (if you set the according preferences), so it can
> act as turn server. stun is really only needed if you want to enforce
> peer-to-peer communication without rtpengine in between.
> 
> Andreas
> 
> On 11/25/2014 08:48 AM, H Yavari wrote:
>> Hi,
>> Thanks for helps. I know that you did all for free. I have a question,
>> Are you using ICE server or STUN? I did all of my test in the local
>> domain and with private IP's.
>> SPCE team, have you any idea for this issue?
>>
>>
>> Regards,
>> H.Yavari
>>
>>
>> ------------------------------------------------------------------------
>> *From:* Nikita Stashkov <snl at sipmobile.org <mailto:snl at sipmobile.org>>
>> **
>> Sorry, I have done all I can do for free. You can test new versions with
>> my site. I think they are working.
>> If you need more help, it can be only commercial support.
>>
>> Regards,
>> Nikita Stashkov
>> 
>>> 24 нояб. 2014 г., в 17:53, H Yavari <hyavari at rocketmail.com
> <mailto:hyavari at rocketmail.com>
>>> <mailto:hyavari at rocketmail.com <mailto:hyavari at rocketmail.com>>>
> написал(а):
>>>
>>>
>>>
>>> Hi,
>>> Very thanks for sharing the script. I'm very confused. I checked the
>>> script line by line and differences are some lines that I think added
>>> in the 3.6.1 and they are not related to the media. I added
>>> "rtcp-mux-demux" flags like your script too. but nothing has changed
>>> and issues not solved.
>>> So I lost my way.  maybe the all problems is from client side. Your
>>> script working with current version of jssip and sipml5? and latest
>>> Chrome and Firefox versions?
>>>
>>> Regards,
>>> H.Yavari
>>>
>>> ------------------------------------------------------------------------
>>> *From:* Nikita Stashkov <snl at sipmobile.org <mailto:snl at sipmobile.org>
> <mailto:snl at sipmobile.org <mailto:snl at sipmobile.org>>>
>>>
>>>
>>> Ok, if it will help you.
>>> Attached is my script (without push), and domain settings.
>>> Should not be understood literally all. I have many changes in config.
>>>
>>>
>>>
>>>
>>>
>>>
>>>> 24 нояб. 2014 г., в 13:07, H Yavari <hyavari at rocketmail.com
> <mailto:hyavari at rocketmail.com>
>>>> <mailto:hyavari at rocketmail.com <mailto:hyavari at rocketmail.com>>>
> написал(а):
>>>>
>>>> Hi,
>>>> Yes, with sipml5 calls have been terminated. I changed all ws to ws
>>>> || wss. I did this too :
>>>>  if(isbflagset(FLB_SAVP_CALLER_SRTP))
>>>>                                {
>>>>                                        xlog("L_INFO", "Try SRTP for
>>>> caller - [% logreq -%]\n");
>>>>                                        $var(rtpp_flags) =
>>>> $var(rtpp_flags) + "SRTP rtcp-mux-demux ";
>>>>                                }
>>>> but did not any changes.
>>>>
>>>> Can you share with me? and you media settings?
>>>>
>>>> I want only use this solution in our website for support calls to our
>>>> IP-PBX.
>>>>
>>>> Thanks.
>>>>
>>>> Regards,
>>>> H.YAvari
>>>> ------------------------------------------------------------------------
>>>> *From:* Nikita Stashkov <snl at sipmobile.org
> <mailto:snl at sipmobile.org> <mailto:snl at sipmobile.org
> <mailto:snl at sipmobile.org>>>
>>>>
>>>>
>>>> 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
> <mailto:hyavari at rocketmail.com>
>>>>> <mailto:hyavari at rocketmail.com <mailto: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
> <mailto:snl at sipmobile.org> <mailto:snl at sipmobile.org
> <mailto: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>
>>>>> <mailto: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> <mailto: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>
>>>>>> <mailto: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> <mailto: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>
>>>>>>>> <mailto: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>
>>>>>>>> <mailto: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/
>>>>>>>> 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>
>>>>>>>>> <mailto: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>
>>>>>>>> <mailto: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
>>>>>>>>>
> ------------------------------------------------------------------------
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Spce-user mailing list
>>>>>>>> Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>
> <mailto:Spce-user at lists.sipwise.com <mailto:Spce-user at lists.sipwise.com>>
>>>>>>>> https://lists.sipwise.com/listinfo/spce-user
> 
> 
> 
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>>
>>
>>
>>
>>
>>
>>
>>
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