[Spce-user] webRTC in production

Tue Dec 2 06:07:56 EST 2014

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
BestSergey

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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