[Spce-user] webRTC in production

Nikita Stashkov snl at sipmobile.org
Tue Nov 25 08:23:49 EST 2014 

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 <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  <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 <mailto: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> <mailto: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>>
> >>> <mailto: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>>
> > <mailto: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>>
> >>>> <mailto: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>> <mailto: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>>
> >>>>> <mailto: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>> <mailto: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>>
> >>>>> <mailto: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>> <mailto: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>>
> >>>>>> <mailto: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>> <mailto: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>>
> >>>>>>>> <mailto: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>>
> >>>>>>>> <mailto: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/ <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>
> > <mailto:odotom at gmail.com <mailto:odotom at gmail.com>>
> >>>>>>>>> <mailto: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>>
> >>>>>>>> <mailto: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>>
> > <mailto: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 <https://lists.sipwise.com/listinfo/spce-user>
> > 
> > 
> > 
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>
> >>>>>>
> >>>>>
> >>>>>
> >>>>
> >>>>
> >>>>
> >>>
> >>>
> >>>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> 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 <https://lists.sipwise.com/listinfo/spce-user>
> >>
> > _______________________________________________
> > 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 <https://lists.sipwise.com/listinfo/spce-user>
> > 
> >
> 
> 

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