[Spce-user] webRTC in production
Nikita Stashkov
snl at sipmobile.org
Mon Nov 24 07:33:09 EST 2014
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> 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>
>
>
> 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>> 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>>
>>
>>
>> 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>> 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>>
>>>
>>>
>>> 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>> 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>>
>>>>
>>>> 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>> 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>>
>>>>>
>>>>> You can try with my site - 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>> 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>>:
>>>>>
>>>>>> 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>
>>>>> https://lists.sipwise.com/listinfo/spce-user <https://lists.sipwise.com/listinfo/spce-user>
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>
>>>
>>
>>
>
>
>
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