[Spce-user] webRTC in production
Sergey Zyrianov
sergey at comoyo.com
Tue Dec 2 06:39:35 EST 2014
Sounds like a NAT breaking your SIP.
Best,
Sergey
On Tue, Dec 2, 2014 at 12:07 PM, H Yavari <hyavari at rocketmail.com> wrote:
> Hi,
> Thanks for reply. sure, and when one side hangs up the call, other side do
> not receive this until 30 sec timeout. I think not any dialog established.
> So problem not be related to SRTP----RTP ???
>
>
> Regards,
> H. Yavari
> ------------------------------
> *From:* Sergey Zyrianov <sergey at comoyo.com>
>
> 30secs teardown usually means missing Ack for Invite
>
> Best
> Sergey
>
> On Tuesday, December 2, 2014, H Yavari <hyavari at rocketmail.com> wrote:
>
>
> Hi,
> I have a question, spce do this SRTP<---->RTP ??
> In my case (webRTC to soft phone) calls, for 30 sec every thing is ok. So
> I think some parameters in SDP is wrong. Have you any idea?
>
> Regards,
> H. Yavari
> ------------------------------
> *From:* H Yavari <hyavari at rocketmail.com>
>
>
> Hi,
> I did my tests with Jitsi soft phone too that support SRTP. But the
> situation not has been changed.
> I'll crazy.
>
> Regards,
> H. Yavari
>
>
>
> ------------------------------
> *From:* Nikita Stashkov <snl at sipmobile.org>
>
> 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> 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> 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
> >>>>>
>
>
>
>
>
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