[Spce-user] webRTC in production
Nikita Stashkov
snl at sipmobile.org
Wed Nov 26 04:02:35 EST 2014
I think it will not work.
You must use SRTP, or modify my config.
Regards,
Nikita Stashkov
> 26 нояб. 2014 г., в 8:52, H Yavari <hyavari at rocketmail.com> написал(а):
>
> Hi,
> I'm using Eyebeam or X-lite and not care about the SRTP. :(
>
> Regards,
> H.Yavari
>
>
> From: Nikita Stashkov <snl at sipmobile.org>
>
>
> 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 <mailto: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 <mailto: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>
>>> >
>>> >
>>> >
>>> >>>>>>>>
>>> >>>>>>>>
>>> >>>>>>>>
>>> >>>>>>>
>>> >>>>>>>
>>> >>>>>>>
>>> >>>>>>
>>> >>>>>>
>>> >>>>>
>>> >>>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>
>>> >>>
>>> >>>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> 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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>
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