<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">In your domain settings do you force SRTP?<div class="">I do.</div><div class=""><br class=""></div><div class="">Regards,</div><div class="">Nikita Stashkov<br class=""><div><blockquote type="cite" class=""><div class="">25 нояб. 2014 г., в 14:08, H Yavari <<a href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>> написал(а):</div><br class="Apple-interchange-newline"><div class=""><div class=""><div style="background-color: rgb(255, 255, 255); font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-size: 13px;" class=""><div dir="ltr" id="yui_3_16_0_1_1416894949617_59109" class=""><span class="">Hi,</span></div><div id="yui_3_16_0_1_1416894949617_59244" dir="ltr" class=""><span id="yui_3_16_0_1_1416894949617_59357" class="">Thanks. I see.</span></div><div id="yui_3_16_0_1_1416894949617_59245" dir="ltr" class=""><span id="yui_3_16_0_1_1416894949617_59356" style="" class="">Have you any idea about this error : "</span><span id="yui_3_16_0_1_1416894949617_59250" class=""><span id="yui_3_16_0_1_1416894949617_59257" style="" class="">SRTP output wanted, but no crypto suite was negotiated</span>" ???</span></div><div id="yui_3_16_0_1_1416894949617_59258" dir="ltr" class="">Is this related to dtls handshake and fingerprints?</div><div id="yui_3_16_0_1_1416894949617_59259" dir="ltr" class="">I see this too: <a id="yui_3_16_0_1_1416894949617_59355" href="https://github.com/sipwise/mediaproxy-ng/blob/master/daemon/rtp.c" class="">https://github.com/sipwise/mediaproxy-ng/blob/master/daemon/rtp.c</a></div><div id="yui_3_16_0_1_1416894949617_59270" dir="ltr" class=""><br class=""></div><div id="yui_3_16_0_1_1416894949617_59352" dir="ltr" class=""><br class=""></div><div id="yui_3_16_0_1_1416894949617_59353" dir="ltr" class="">Regards,</div><div id="yui_3_16_0_1_1416894949617_59354" dir="ltr" class="">H. Yavari<br class=""></div><div id="yui_3_16_0_1_1416894949617_59049" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;" class=""> <div id="yui_3_16_0_1_1416894949617_59048" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;" class=""> <div id="yui_3_16_0_1_1416894949617_59107" class="y_msg_container"> <hr id="yui_3_16_0_1_1416894949617_59108" size="1" class=""><div id="yui_3_16_0_1_1416894949617_59260" class=""> <font id="yui_3_16_0_1_1416894949617_59046" face="Arial" size="2" class=""> <b id="yui_3_16_0_1_1416894949617_59106" class=""><span id="yui_3_16_0_1_1416894949617_59105" style="font-weight:bold;" class="">From:</span></b> Andreas Granig <<a href="mailto:agranig@sipwise.com" class="">agranig@sipwise.com</a>></font></div><div id="yui_3_16_0_1_1416894949617_59261" class=""><br class=""><font id="yui_3_16_0_1_1416894949617_59046" face="Arial" size="2" class=""></font></div><div id="yui_3_16_0_1_1416894949617_59263" dir="ltr" class="">Please see <a id="yui_3_16_0_1_1416894949617_59262" href="https://github.com/sipwise/rtpengine#offer-message" target="_blank" class="">https://github.com/sipwise/rtpengine#offer-message </a>for<br class=""></div><div id="yui_3_16_0_1_1416894949617_59264" dir="ltr" class="">available options and their possible values.<br class=""></div><div id="yui_3_16_0_1_1416894949617_59265" dir="ltr" class=""><br class=""></div><div id="yui_3_16_0_1_1416894949617_59266" dir="ltr" class="">Andreas<br class=""></div><div dir="ltr" class=""><br class=""></div><div dir="ltr" class="">On 11/25/2014 01:35 PM, H Yavari wrote:<br class=""></div><div dir="ltr" class="">> Hi,<br class=""></div><div dir="ltr" class="">> I copied the all flags same as Nikita script.Nothing has changed but in<br class=""></div><div dir="ltr" class="">> the rtp.log there are some lines :<br class=""></div><div dir="ltr" class="">> Nov 24 07:36:55 spce rtpengine[6426]: Unknown flag encountered: 'symmetric'<br class=""></div><div dir="ltr" class="">> Nov 24 07:36:55 spce rtpengine[6426]: Unknown 'rtcp-mux' flag<br class=""></div><div dir="ltr" class="">> encountered: 'demuxSRTP'<br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> Nov 24 07:37:00 spce rtpengine[6426]:<br class=""></div><div dir="ltr" class="">> [f5008c55-8329-f08e-e024-81d8260b1708 port 30865] SRTCP output wanted,<br class=""></div><div dir="ltr" class="">> but no crypto suite was negotiated<br class=""></div><div dir="ltr" class="">> .<br class=""></div><div dir="ltr" class="">> .<br class=""></div><div dir="ltr" class="">> .<br class=""></div><div dir="ltr" class="">> .<br class=""></div><div dir="ltr" class="">> Nov 24 07:37:32 spce rtpengine[6426]:<br class=""></div><div dir="ltr" class="">> [f5008c55-8329-f08e-e024-81d8260b1708] Scheduling deletion of call<br class=""></div><div dir="ltr" class="">> branch 'R7t3SMDI7STFq4A53a9w' in 30 seconds<br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> this flags not supported by rtpengine now? (3.6.1)<br class=""></div><div dir="ltr" class="">> how suite crypto will be negotiated?<br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> Regards,<br class=""></div><div dir="ltr" class="">> H. Yavari<br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> ------------------------------------------------------------------------<br class=""></div><div dir="ltr" class="">> *From:* Andreas Granig <<a ymailto="mailto:agranig@sipwise.com" href="mailto:agranig@sipwise.com" class="">agranig@sipwise.com</a>><br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> You don't need stun/turn with rtpengine, because it puts itself into the<br class=""></div><div dir="ltr" class="">> SDP as ICE candidate (if you set the according preferences), so it can<br class=""></div><div dir="ltr" class="">> act as turn server. stun is really only needed if you want to enforce<br class=""></div><div dir="ltr" class="">> peer-to-peer communication without rtpengine in between.<br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> Andreas<br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> On 11/25/2014 08:48 AM, H Yavari wrote:<br class=""></div><div dir="ltr" class="">>> Hi,<br class=""></div><div dir="ltr" class="">>> Thanks for helps. I know that you did all for free. I have a question,<br class=""></div><div dir="ltr" class="">>> Are you using ICE server or STUN? I did all of my test in the local<br class=""></div><div dir="ltr" class="">>> domain and with private IP's.<br class=""></div><div dir="ltr" class="">>> SPCE team, have you any idea for this issue?<br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>> Regards,<br class=""></div><div dir="ltr" class="">>> H.Yavari<br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>> ------------------------------------------------------------------------<br class=""></div><div dir="ltr" class="">>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>>><br class=""></div><div dir="ltr" class="">>> **<br class=""></div><div dir="ltr" class="">>> Sorry, I have done all I can do for free. You can test new versions with<br class=""></div><div dir="ltr" class="">>> my site. I think they are working.<br class=""></div><div dir="ltr" class="">>> If you need more help, it can be only commercial support.<br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>> Regards,<br class=""></div><div dir="ltr" class="">>> Nikita Stashkov<br class=""></div><div dir="ltr" class="">>> <br class=""></div><div dir="ltr" class="">>>> 24 нояб. 2014 г., в 17:53, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>><br class=""></div><div dir="ltr" class="">>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>>>><br class=""></div><div dir="ltr" class="">> написал(а):<br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>> Hi,<br class=""></div><div dir="ltr" class="">>>> Very thanks for sharing the script. I'm very confused. I checked the<br class=""></div><div dir="ltr" class="">>>> script line by line and differences are some lines that I think added<br class=""></div><div dir="ltr" class="">>>> in the 3.6.1 and they are not related to the media. I added<br class=""></div><div dir="ltr" class="">>>> "rtcp-mux-demux" flags like your script too. but nothing has changed<br class=""></div><div dir="ltr" class="">>>> and issues not solved.<br class=""></div><div dir="ltr" class="">>>> So I lost my way. maybe the all problems is from client side. Your<br class=""></div><div dir="ltr" class="">>>> script working with current version of jssip and sipml5? and latest<br class=""></div><div dir="ltr" class="">>>> Chrome and Firefox versions?<br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>> Regards,<br class=""></div><div dir="ltr" class="">>>> H.Yavari<br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>> ------------------------------------------------------------------------<br class=""></div><div dir="ltr" class="">>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>> Ok, if it will help you.<br class=""></div><div dir="ltr" class="">>>> Attached is my script (without push), and domain settings.<br class=""></div><div dir="ltr" class="">>>> Should not be understood literally all. I have many changes in config.<br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>>> 24 нояб. 2014 г., в 13:07, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>><br class=""></div><div dir="ltr" class="">>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>>>><br class=""></div><div dir="ltr" class="">> написал(а):<br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>> Hi,<br class=""></div><div dir="ltr" class="">>>>> Yes, with sipml5 calls have been terminated. I changed all ws to ws<br class=""></div><div dir="ltr" class="">>>>> || wss. I did this too :<br class=""></div><div dir="ltr" class="">>>>> if(isbflagset(FLB_SAVP_CALLER_SRTP))<br class=""></div><div dir="ltr" class="">>>>> {<br class=""></div><div dir="ltr" class="">>>>> xlog("L_INFO", "Try SRTP for<br class=""></div><div dir="ltr" class="">>>>> caller - [% logreq -%]\n");<br class=""></div><div dir="ltr" class="">>>>> $var(rtpp_flags) =<br class=""></div><div dir="ltr" class="">>>>> $var(rtpp_flags) + "SRTP rtcp-mux-demux ";<br class=""></div><div dir="ltr" class="">>>>> }<br class=""></div><div dir="ltr" class="">>>>> but did not any changes.<br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>> Can you share with me? and you media settings?<br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>> I want only use this solution in our website for support calls to our<br class=""></div><div dir="ltr" class="">>>>> IP-PBX.<br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>> Thanks.<br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>> H.YAvari<br class=""></div><div dir="ltr" class="">>>>> ------------------------------------------------------------------------<br class=""></div><div dir="ltr" class="">>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>>>><br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>> And the first one is sipml5?<br class=""></div><div dir="ltr" class="">>>>> In my config both are working.<br class=""></div><div dir="ltr" class="">>>>> Check again your script. There is not one place, where automatic<br class=""></div><div dir="ltr" class="">>>>> detection is done.<br class=""></div><div dir="ltr" class="">>>>> I don’t exactly remember. It was about 4-5 month ago. But I think,<br class=""></div><div dir="ltr" class="">>>>> difference is between ws and wss.<br class=""></div><div dir="ltr" class="">>>>> Sorry, I can not publish my script. There are many other things,<br class=""></div><div dir="ltr" class="">>>>> including push notifications.<br class=""></div><div dir="ltr" class="">>>>> I think Sipwise will be not happy, if I publish it.<br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>> Nikita Stashkov<br class=""></div><div dir="ltr" class="">>>>>> 24 нояб. 2014 г., в 11:45, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>><br class=""></div><div dir="ltr" class="">>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>>>><br class=""></div><div dir="ltr" class="">> написал(а):<br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>> Hi,<br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>> I noticed a new thing that when I using jssip, calls not terminated.<br class=""></div><div dir="ltr" class="">>>>>> in the logs and I didn't see any rtcp-mux. so this two webRTC client<br class=""></div><div dir="ltr" class="">>>>>> is different in using SDP params?<br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>>> H.Yavari<br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">> ------------------------------------------------------------------------<br class=""></div><div dir="ltr" class="">>>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>> There may be different logic. I am doing it if caller is WebRTC, and<br class=""></div><div dir="ltr" class="">>>>>> callee is SIP.<br class=""></div><div dir="ltr" class="">>>>>> Simply add this flag, when calling Rtpengine, like all other flags.<br class=""></div><div dir="ltr" class="">>>>>> You can do nothing if both are WebRTC.<br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>>> Nikita Stashkov<br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>> 24 нояб. 2014 г., в 11:13, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>><br class=""></div><div dir="ltr" class="">>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>>>><br class=""></div><div dir="ltr" class="">> написал(а):<br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>>> Hi,<br class=""></div><div dir="ltr" class="">>>>>>> I checked it. the client (webRTC browser-sipml5) send rtp-mux. and<br class=""></div><div dir="ltr" class="">>>>>>> is in the rtp.log too. so how can I disable this? or how can I add<br class=""></div><div dir="ltr" class="">>>>>>> rtcp-mux-demux ? I should do this for all calls? or only for webRTC<br class=""></div><div dir="ltr" class="">>>>>>> client?<br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>> Thanks a lot.<br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>>>> H.Yavari<br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">> ------------------------------------------------------------------------<br class=""></div><div dir="ltr" class="">>>>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>>>><br class=""></div><div dir="ltr" class="">>>>>>> **<br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>> You need to look logs from WebRTC client and Pcap from SIP.<br class=""></div><div dir="ltr" class="">>>>>>> Of cource, if SIP client recives SDP with rtp-mux, he will not<br class=""></div><div dir="ltr" class="">>>>>>> understand it. And after 30 sec call will be terminated. But you<br class=""></div><div dir="ltr" class="">>>>>>> must see logs. My system is based on SPCE 3.2, and manually<br class=""></div><div dir="ltr" class="">>>>>>> compiled rtpengine. And I don't know was changed in current<br class=""></div><div dir="ltr" class="">>>>>>> version. Also, you can look rtp.log. Sometimes it helps.<br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>>>> Nikita Stashkov<br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>> 23. nov. 2014, в 13.40, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>><br class=""></div><div dir="ltr" class="">>>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>>>><br class=""></div><div dir="ltr" class="">> написал(а):<br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>> Hi,<br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>> Dear I did this before that I changed "ws" with "wss" but now<br class=""></div><div dir="ltr" class="">>>>>>>> after your reply I did "ws" || "wss". but not any changes.<br class=""></div><div dir="ltr" class="">>>>>>>> As I told before, now my main problem is calls hangup after 30<br class=""></div><div dir="ltr" class="">>>>>>>> sec. In your opinion the rtcp-mux-demux flags adding will solve this?<br class=""></div><div dir="ltr" class="">>>>>>>> another point is that before 30 sec, If any call parties (caller:<br class=""></div><div dir="ltr" class="">>>>>>>> browser and callee: soft phone) hangs up, the call not terminate<br class=""></div><div dir="ltr" class="">>>>>>>> until 30 sec timeout. I think that the dialog of a call not<br class=""></div><div dir="ltr" class="">>>>>>>> recognized.<br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>> So situation is complicated :)<br class=""></div><div dir="ltr" class="">>>>>>>> SPCE specialist plz help!<br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>>>>> H. Yavari<br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">> ------------------------------------------------------------------------<br class=""></div><div dir="ltr" class="">>>>>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>>>><br class=""></div><div dir="ltr" class="">>>>>>>> **<br class=""></div><div dir="ltr" class="">>>>>>>> Sorry, I can not share my script.<br class=""></div><div dir="ltr" class="">>>>>>>> What can you do.<br class=""></div><div dir="ltr" class="">>>>>>>> Look the<br class=""></div><div dir="ltr" class="">>>>>>>> script /etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.tt2<br class=""></div><div dir="ltr" class="">>>>>>>> Of course, before modifying copy it to proxy.cfg.customtt.tt2<br class=""></div><div dir="ltr" class="">>>>>>>> I think, webrtc endpoint automatic detection is not working for you.<br class=""></div><div dir="ltr" class="">>>>>>>> It must look like this:<br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>> if($(ru{uri.param,transport}) == "ws" ||<br class=""></div><div dir="ltr" class="">>>>>>>> $(ru{uri.param,transport}) == "wss»)<br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>> Then check flags you are sending to rtpengine.<br class=""></div><div dir="ltr" class="">>>>>>>> To call SIP clients you must use flag rtcp-mux-demux<br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>>>>> Nikita Stashkov<br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> 22 нояб. 2014 г., в 20:28, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>><br class=""></div><div dir="ltr" class="">>>>>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>>>><br class=""></div><div dir="ltr" class="">> написал(а):<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> Hi,<br class=""></div><div dir="ltr" class="">>>>>>>>> I checked you site. it seems that is a good webRTC solution.<br class=""></div><div dir="ltr" class="">>>>>>>>> Can you share with us your experience to solve our problem? or<br class=""></div><div dir="ltr" class="">>>>>>>>> any script modifications?<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>>>>>> H.Yavari<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">> ------------------------------------------------------------------------<br class=""></div><div dir="ltr" class="">>>>>>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>><br class=""></div><div dir="ltr" class="">>>>>>>>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org" class="">snl@sipmobile.org</a>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> You can try with my site - <a href="http://www.sipmobile.org" class="">www.sipmobile.org</a><br class=""></div><div dir="ltr" class="">>>>>>>>> <<a href="http://www.sipmobile.org/" target="_blank" class="">http://www.sipmobile.org/</a>>.<br class=""></div><div dir="ltr" class="">>>>>>>>> Create accounts: <a href="https://www.sipmobile.org/register/" target="_blank" class="">https://www.sipmobile.org/register/</a><br class=""></div><div dir="ltr" class="">>>>>>>>> And try to call with webRTC client and SIP.<br class=""></div><div dir="ltr" class="">>>>>>>>> I have modified some Kamailio SPCE scripts.<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>>>>>> Nikita Stashkov<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>>> 22 нояб. 2014 г., в 16:04, Thomas Odorfer <<a ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com" class="">odotom@gmail.com</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com" class="">odotom@gmail.com</a>><br class=""></div><div dir="ltr" class="">>>>>>>>>> <mailto:<a ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com" class="">odotom@gmail.com</a> <mailto:<a ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com" class="">odotom@gmail.com</a>>>> написал(а):<br class=""></div><div dir="ltr" class="">>>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> Hi,<br class=""></div><div dir="ltr" class="">>>>>>>>> not sure if I understood correctly which scenario works and which<br class=""></div><div dir="ltr" class="">>>>>>>>> not.<br class=""></div><div dir="ltr" class="">>>>>>>>> So browser to soft phone is now working, but what is the meaning<br class=""></div><div dir="ltr" class="">>>>>>>>> of browser to client? Which client?<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> I tested myself and I have to confess that I had to do some<br class=""></div><div dir="ltr" class="">>>>>>>>> changes in the account configs for soft phones where I am not<br class=""></div><div dir="ltr" class="">>>>>>>>> happy about.<br class=""></div><div dir="ltr" class="">>>>>>>>> It only worked between browser-webrtc and soft phone when the<br class=""></div><div dir="ltr" class="">>>>>>>>> corresponding account for the soft phone - nat & media flow<br class=""></div><div dir="ltr" class="">>>>>>>>> control had been changed to "force avp"“ and "force rtp“ ie. no<br class=""></div><div dir="ltr" class="">>>>>>>>> encryption.<br class=""></div><div dir="ltr" class="">>>>>>>>> (I have to investigate that one - could be related to an upgrade<br class=""></div><div dir="ltr" class="">>>>>>>>> I had performed last week - usually srtp should also work with<br class=""></div><div dir="ltr" class="">>>>>>>>> softphones, within the ftp.log there was „SRTP output wanted but<br class=""></div><div dir="ltr" class="">>>>>>>>> no crypto suite was negotiated“).<br class=""></div><div dir="ltr" class="">>>>>>>>> However, after my changes the following tests had been successful:<br class=""></div><div dir="ltr" class="">>>>>>>>> browser webrtc to softphone (eg. jitsi, counterpath x-lite -<br class=""></div><div dir="ltr" class="">>>>>>>>> should be software compatible with eyebeam)<br class=""></div><div dir="ltr" class="">>>>>>>>> browser webrtc to another browser webrtc (jssip-0.50)<br class=""></div><div dir="ltr" class="">>>>>>>>> browser webrtc to pstn via sip trunking (standard sip trunk,<br class=""></div><div dir="ltr" class="">>>>>>>>> peer settings for media force „rtp“, „force rtp“, „always with<br class=""></div><div dir="ltr" class="">>>>>>>>> plain SDP“)<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> That is based on the latest SPCE version 3.6.1.<br class=""></div><div dir="ltr" class="">>>>>>>>> What does not seem to be achievable at the moment that you can<br class=""></div><div dir="ltr" class="">>>>>>>>> have an account that supports „standard“ and webrtc<br class=""></div><div dir="ltr" class="">>>>>>>>> simultaneously ( at least I haven’t succeeded with such a setup,<br class=""></div><div dir="ltr" class="">>>>>>>>> maybe some sipwise/kamailio/rtpengine expert knows the trick).<br class=""></div><div dir="ltr" class="">>>>>>>>> And I do not have a solution yet how to share one phone number<br class=""></div><div dir="ltr" class="">>>>>>>>> between two accounts with different profiles.<br class=""></div><div dir="ltr" class="">>>>>>>>> The only solution I have at the moment is that I put a webrtc<br class=""></div><div dir="ltr" class="">>>>>>>>> gateway (similar to webrtc2sip from doubango) in front of SPCE<br class=""></div><div dir="ltr" class="">>>>>>>>> for webrtc clients.<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> For your particular problem, maybe you have to check whether your<br class=""></div><div dir="ltr" class="">>>>>>>>> domain settings allow „bypass rtp proxy“ behind the same NAT -<br class=""></div><div dir="ltr" class="">>>>>>>>> assuming you are testing wthin your LAN - this should be set to<br class=""></div><div dir="ltr" class="">>>>>>>>> never.<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> Good luck<br class=""></div><div dir="ltr" class="">>>>>>>>> Thomas<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> Am 22.11.2014 um 12:49 schrieb H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>><br class=""></div><div dir="ltr" class="">>>>>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com" class="">hyavari@rocketmail.com</a>>>>:<br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>>> Hi,<br class=""></div><div dir="ltr" class="">>>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>>> I did this configs:<br class=""></div><div dir="ltr" class="">>>>>>>>>> use_rtpproxy: „Always with rtpptoxy as only ICE candidate“<br class=""></div><div dir="ltr" class="">>>>>>>>>> rtcp_feedback: „Force AVP“<br class=""></div><div dir="ltr" class="">>>>>>>>>> srtp_transcoding: „Force RTP“<br class=""></div><div dir="ltr" class="">>>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>>> now calls between browser to soft phone is ok, but browser to<br class=""></div><div dir="ltr" class="">>>>>>>>>> client and browser to browser receive this error "Failed to get<br class=""></div><div dir="ltr" class="">>>>>>>>>> local SDP"<br class=""></div><div dir="ltr" class="">>>>>>>>>> and calls not be established. Have you any idea about this<br class=""></div><div dir="ltr" class="">>>>>>>>>> situation?<br class=""></div><div dir="ltr" class="">>>>>>>>>> Thanks for helps.<br class=""></div><div dir="ltr" class="">>>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>>> Regards,<br class=""></div><div dir="ltr" class="">>>>>>>>>> H.Yavari<br class=""></div><div dir="ltr" class="">>>>>>>>>><br class=""></div><div dir="ltr" class="">> ------------------------------------------------------------------------<br class=""></div><div dir="ltr" class="">>>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>> _______________________________________________<br class=""></div><div dir="ltr" class="">>>>>>>>> Spce-user mailing list<br class=""></div><div dir="ltr" class="">>>>>>>>> <a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com" class="">Spce-user@lists.sipwise.com</a> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com" class="">Spce-user@lists.sipwise.com</a>><br class=""></div><div dir="ltr" class="">> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com" class="">Spce-user@lists.sipwise.com</a> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com" class="">Spce-user@lists.sipwise.com</a>>><br class=""></div><div dir="ltr" class="">>>>>>>>> <a href="https://lists.sipwise.com/listinfo/spce-user" target="_blank" class="">https://lists.sipwise.com/listinfo/spce-user</a><br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>>><br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>>><br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>>><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">>> _______________________________________________<br class=""></div><div dir="ltr" class="">>> Spce-user mailing list<br class=""></div><div dir="ltr" class="">>> <a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com" class="">Spce-user@lists.sipwise.com</a> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com" class="">Spce-user@lists.sipwise.com</a>><br class=""></div><div dir="ltr" class="">>> <a href="https://lists.sipwise.com/listinfo/spce-user" target="_blank" class="">https://lists.sipwise.com/listinfo/spce-user</a><br class=""></div><div dir="ltr" class="">>><br class=""></div><div dir="ltr" class="">> _______________________________________________<br class=""></div><div dir="ltr" class="">> Spce-user mailing list<br class=""></div><div dir="ltr" class="">> <a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com" class="">Spce-user@lists.sipwise.com</a> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com" class="">Spce-user@lists.sipwise.com</a>><br class=""></div><div dir="ltr" class="">> <a href="https://lists.sipwise.com/listinfo/spce-user" target="_blank" class="">https://lists.sipwise.com/listinfo/spce-user</a><br class=""></div><div dir="ltr" class="">> <br class=""></div><div dir="ltr" class="">> </div><br class=""><br class=""></div> </div> </div> </div></div></div></blockquote></div><br class=""></div></body></html>