[Spce-user] webRTC in production
Nikita Stashkov
snl at sipmobile.org
Wed Nov 26 02:37:15 EST 2014
Yes, it is write.
And did you with on SRTP on your phone?
Regards,
Nikita Stashkov
> 25 нояб. 2014 г., в 14:42, H Yavari <hyavari at rocketmail.com> написал(а):
>
> 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 <mailto: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 <mailto: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/ <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>
>> >
>> >
>> >
>> >>>>>>>>
>> >>>>>>>>
>> >>>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>
>> >>>>>>
>> >>>>>
>> >>>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>
>> >>>
>> >>>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> _______________________________________________
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>> >> 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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>> > https://lists.sipwise.com/listinfo/spce-user <https://lists.sipwise.com/listinfo/spce-user>
>> >
>> >
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