<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px"><div dir="ltr" id="yui_3_16_0_1_1416894949617_59109"><span>Hi,</span></div><div id="yui_3_16_0_1_1416894949617_59244" dir="ltr"><span id="yui_3_16_0_1_1416894949617_59357">Thanks. I see.</span></div><div id="yui_3_16_0_1_1416894949617_59245" dir="ltr"><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"><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">Is this related to dtls handshake and fingerprints?</div><div id="yui_3_16_0_1_1416894949617_59259" dir="ltr">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">https://github.com/sipwise/mediaproxy-ng/blob/master/daemon/rtp.c</a></div><div id="yui_3_16_0_1_1416894949617_59270" dir="ltr"><br></div><div id="yui_3_16_0_1_1416894949617_59352" dir="ltr"><br></div><div id="yui_3_16_0_1_1416894949617_59353" dir="ltr">Regards,</div><div id="yui_3_16_0_1_1416894949617_59354" dir="ltr">H. Yavari<br></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;"> <div id="yui_3_16_0_1_1416894949617_59048" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div id="yui_3_16_0_1_1416894949617_59107" class="y_msg_container"> <hr id="yui_3_16_0_1_1416894949617_59108" size="1"><div id="yui_3_16_0_1_1416894949617_59260"> <font id="yui_3_16_0_1_1416894949617_59046" face="Arial" size="2"> <b id="yui_3_16_0_1_1416894949617_59106"><span id="yui_3_16_0_1_1416894949617_59105" style="font-weight:bold;">From:</span></b> Andreas Granig <agranig@sipwise.com></font></div><div id="yui_3_16_0_1_1416894949617_59261"><br><font id="yui_3_16_0_1_1416894949617_59046" face="Arial" size="2"></font></div><div id="yui_3_16_0_1_1416894949617_59263" dir="ltr">Please see <a id="yui_3_16_0_1_1416894949617_59262" href="https://github.com/sipwise/rtpengine#offer-message" target="_blank">https://github.com/sipwise/rtpengine#offer-message </a>for<br></div><div id="yui_3_16_0_1_1416894949617_59264" dir="ltr">available options and their possible values.<br></div><div id="yui_3_16_0_1_1416894949617_59265" dir="ltr"><br></div><div id="yui_3_16_0_1_1416894949617_59266" dir="ltr">Andreas<br></div><div dir="ltr"><br></div><div dir="ltr">On 11/25/2014 01:35 PM, H Yavari wrote:<br></div><div dir="ltr">> Hi,<br></div><div dir="ltr">> I copied the all flags same as Nikita script.Nothing has changed but in<br></div><div dir="ltr">> the rtp.log there are some lines :<br></div><div dir="ltr">> Nov 24 07:36:55 spce rtpengine[6426]: Unknown flag encountered: 'symmetric'<br></div><div dir="ltr">> Nov 24 07:36:55 spce rtpengine[6426]: Unknown 'rtcp-mux' flag<br></div><div dir="ltr">> encountered: 'demuxSRTP'<br></div><div dir="ltr">> <br></div><div dir="ltr">> Nov 24 07:37:00 spce rtpengine[6426]:<br></div><div dir="ltr">> [f5008c55-8329-f08e-e024-81d8260b1708 port 30865] SRTCP output wanted,<br></div><div dir="ltr">> but no crypto suite was negotiated<br></div><div dir="ltr">> .<br></div><div dir="ltr">> .<br></div><div dir="ltr">> .<br></div><div dir="ltr">> .<br></div><div dir="ltr">> Nov 24 07:37:32 spce rtpengine[6426]:<br></div><div dir="ltr">> [f5008c55-8329-f08e-e024-81d8260b1708] Scheduling deletion of call<br></div><div dir="ltr">> branch 'R7t3SMDI7STFq4A53a9w' in 30 seconds<br></div><div dir="ltr">> <br></div><div dir="ltr">> this flags not supported by rtpengine now? (3.6.1)<br></div><div dir="ltr">> how suite crypto will be negotiated?<br></div><div dir="ltr">> <br></div><div dir="ltr">> <br></div><div dir="ltr">> Regards,<br></div><div dir="ltr">> H. Yavari<br></div><div dir="ltr">> <br></div><div dir="ltr">> <br></div><div dir="ltr">> ------------------------------------------------------------------------<br></div><div dir="ltr">> *From:* Andreas Granig <<a ymailto="mailto:agranig@sipwise.com" href="mailto:agranig@sipwise.com">agranig@sipwise.com</a>><br></div><div dir="ltr">> <br></div><div dir="ltr">> <br></div><div dir="ltr">> You don't need stun/turn with rtpengine, because it puts itself into the<br></div><div dir="ltr">> SDP as ICE candidate (if you set the according preferences), so it can<br></div><div dir="ltr">> act as turn server. stun is really only needed if you want to enforce<br></div><div dir="ltr">> peer-to-peer communication without rtpengine in between.<br></div><div dir="ltr">> <br></div><div dir="ltr">> Andreas<br></div><div dir="ltr">> <br></div><div dir="ltr">> On 11/25/2014 08:48 AM, H Yavari wrote:<br></div><div dir="ltr">>> Hi,<br></div><div dir="ltr">>> Thanks for helps. I know that you did all for free. I have a question,<br></div><div dir="ltr">>> Are you using ICE server or STUN? I did all of my test in the local<br></div><div dir="ltr">>> domain and with private IP's.<br></div><div dir="ltr">>> SPCE team, have you any idea for this issue?<br></div><div dir="ltr">>><br></div><div dir="ltr">>><br></div><div dir="ltr">>> Regards,<br></div><div dir="ltr">>> H.Yavari<br></div><div dir="ltr">>><br></div><div dir="ltr">>><br></div><div dir="ltr">>> ------------------------------------------------------------------------<br></div><div dir="ltr">>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>><br></div><div dir="ltr">>> **<br></div><div dir="ltr">>> Sorry, I have done all I can do for free. You can test new versions with<br></div><div dir="ltr">>> my site. I think they are working.<br></div><div dir="ltr">>> If you need more help, it can be only commercial support.<br></div><div dir="ltr">>><br></div><div dir="ltr">>> Regards,<br></div><div dir="ltr">>> Nikita Stashkov<br></div><div dir="ltr">>> <br></div><div dir="ltr">>>> 24 нояб. 2014 г., в 17:53, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>><br></div><div dir="ltr">>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>>><br></div><div dir="ltr">> написал(а):<br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>> Hi,<br></div><div dir="ltr">>>> Very thanks for sharing the script. I'm very confused. I checked the<br></div><div dir="ltr">>>> script line by line and differences are some lines that I think added<br></div><div dir="ltr">>>> in the 3.6.1 and they are not related to the media. I added<br></div><div dir="ltr">>>> "rtcp-mux-demux" flags like your script too. but nothing has changed<br></div><div dir="ltr">>>> and issues not solved.<br></div><div dir="ltr">>>> So I lost my way. maybe the all problems is from client side. Your<br></div><div dir="ltr">>>> script working with current version of jssip and sipml5? and latest<br></div><div dir="ltr">>>> Chrome and Firefox versions?<br></div><div dir="ltr">>>><br></div><div dir="ltr">>>> Regards,<br></div><div dir="ltr">>>> H.Yavari<br></div><div dir="ltr">>>><br></div><div dir="ltr">>>> ------------------------------------------------------------------------<br></div><div dir="ltr">>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>> Ok, if it will help you.<br></div><div dir="ltr">>>> Attached is my script (without push), and domain settings.<br></div><div dir="ltr">>>> Should not be understood literally all. I have many changes in config.<br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>>> 24 нояб. 2014 г., в 13:07, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>><br></div><div dir="ltr">>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>>><br></div><div dir="ltr">> написал(а):<br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>> Hi,<br></div><div dir="ltr">>>>> Yes, with sipml5 calls have been terminated. I changed all ws to ws<br></div><div dir="ltr">>>>> || wss. I did this too :<br></div><div dir="ltr">>>>> if(isbflagset(FLB_SAVP_CALLER_SRTP))<br></div><div dir="ltr">>>>> {<br></div><div dir="ltr">>>>> xlog("L_INFO", "Try SRTP for<br></div><div dir="ltr">>>>> caller - [% logreq -%]\n");<br></div><div dir="ltr">>>>> $var(rtpp_flags) =<br></div><div dir="ltr">>>>> $var(rtpp_flags) + "SRTP rtcp-mux-demux ";<br></div><div dir="ltr">>>>> }<br></div><div dir="ltr">>>>> but did not any changes.<br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>> Can you share with me? and you media settings?<br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>> I want only use this solution in our website for support calls to our<br></div><div dir="ltr">>>>> IP-PBX.<br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>> Thanks.<br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>> Regards,<br></div><div dir="ltr">>>>> H.YAvari<br></div><div dir="ltr">>>>> ------------------------------------------------------------------------<br></div><div dir="ltr">>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>>><br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>> And the first one is sipml5?<br></div><div dir="ltr">>>>> In my config both are working.<br></div><div dir="ltr">>>>> Check again your script. There is not one place, where automatic<br></div><div dir="ltr">>>>> detection is done.<br></div><div dir="ltr">>>>> I don’t exactly remember. It was about 4-5 month ago. But I think,<br></div><div dir="ltr">>>>> difference is between ws and wss.<br></div><div dir="ltr">>>>> Sorry, I can not publish my script. There are many other things,<br></div><div dir="ltr">>>>> including push notifications.<br></div><div dir="ltr">>>>> I think Sipwise will be not happy, if I publish it.<br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>> Regards,<br></div><div dir="ltr">>>>> Nikita Stashkov<br></div><div dir="ltr">>>>>> 24 нояб. 2014 г., в 11:45, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>><br></div><div dir="ltr">>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>>><br></div><div dir="ltr">> написал(а):<br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>> Hi,<br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>> I noticed a new thing that when I using jssip, calls not terminated.<br></div><div dir="ltr">>>>>> in the logs and I didn't see any rtcp-mux. so this two webRTC client<br></div><div dir="ltr">>>>>> is different in using SDP params?<br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>> Regards,<br></div><div dir="ltr">>>>>> H.Yavari<br></div><div dir="ltr">>>>>><br></div><div dir="ltr">> ------------------------------------------------------------------------<br></div><div dir="ltr">>>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>> There may be different logic. I am doing it if caller is WebRTC, and<br></div><div dir="ltr">>>>>> callee is SIP.<br></div><div dir="ltr">>>>>> Simply add this flag, when calling Rtpengine, like all other flags.<br></div><div dir="ltr">>>>>> You can do nothing if both are WebRTC.<br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>> Regards,<br></div><div dir="ltr">>>>>> Nikita Stashkov<br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>> 24 нояб. 2014 г., в 11:13, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>><br></div><div dir="ltr">>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>>><br></div><div dir="ltr">> написал(а):<br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>>> Hi,<br></div><div dir="ltr">>>>>>> I checked it. the client (webRTC browser-sipml5) send rtp-mux. and<br></div><div dir="ltr">>>>>>> is in the rtp.log too. so how can I disable this? or how can I add<br></div><div dir="ltr">>>>>>> rtcp-mux-demux ? I should do this for all calls? or only for webRTC<br></div><div dir="ltr">>>>>>> client?<br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>> Thanks a lot.<br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>> Regards,<br></div><div dir="ltr">>>>>>> H.Yavari<br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">> ------------------------------------------------------------------------<br></div><div dir="ltr">>>>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>>><br></div><div dir="ltr">>>>>>> **<br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>> You need to look logs from WebRTC client and Pcap from SIP.<br></div><div dir="ltr">>>>>>> Of cource, if SIP client recives SDP with rtp-mux, he will not<br></div><div dir="ltr">>>>>>> understand it. And after 30 sec call will be terminated. But you<br></div><div dir="ltr">>>>>>> must see logs. My system is based on SPCE 3.2, and manually<br></div><div dir="ltr">>>>>>> compiled rtpengine. And I don't know was changed in current<br></div><div dir="ltr">>>>>>> version. Also, you can look rtp.log. Sometimes it helps.<br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>> Regards,<br></div><div dir="ltr">>>>>>> Nikita Stashkov<br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>> 23. nov. 2014, в 13.40, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>><br></div><div dir="ltr">>>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>>><br></div><div dir="ltr">> написал(а):<br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>>> Hi,<br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>> Dear I did this before that I changed "ws" with "wss" but now<br></div><div dir="ltr">>>>>>>> after your reply I did "ws" || "wss". but not any changes.<br></div><div dir="ltr">>>>>>>> As I told before, now my main problem is calls hangup after 30<br></div><div dir="ltr">>>>>>>> sec. In your opinion the rtcp-mux-demux flags adding will solve this?<br></div><div dir="ltr">>>>>>>> another point is that before 30 sec, If any call parties (caller:<br></div><div dir="ltr">>>>>>>> browser and callee: soft phone) hangs up, the call not terminate<br></div><div dir="ltr">>>>>>>> until 30 sec timeout. I think that the dialog of a call not<br></div><div dir="ltr">>>>>>>> recognized.<br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>> So situation is complicated :)<br></div><div dir="ltr">>>>>>>> SPCE specialist plz help!<br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>> Regards,<br></div><div dir="ltr">>>>>>>> H. Yavari<br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">> ------------------------------------------------------------------------<br></div><div dir="ltr">>>>>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>>><br></div><div dir="ltr">>>>>>>> **<br></div><div dir="ltr">>>>>>>> Sorry, I can not share my script.<br></div><div dir="ltr">>>>>>>> What can you do.<br></div><div dir="ltr">>>>>>>> Look the<br></div><div dir="ltr">>>>>>>> script /etc/ngcp-config/templates/etc/kamailio/proxy/proxy.cfg.tt2<br></div><div dir="ltr">>>>>>>> Of course, before modifying copy it to proxy.cfg.customtt.tt2<br></div><div dir="ltr">>>>>>>> I think, webrtc endpoint automatic detection is not working for you.<br></div><div dir="ltr">>>>>>>> It must look like this:<br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>> if($(ru{uri.param,transport}) == "ws" ||<br></div><div dir="ltr">>>>>>>> $(ru{uri.param,transport}) == "wss»)<br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>> Then check flags you are sending to rtpengine.<br></div><div dir="ltr">>>>>>>> To call SIP clients you must use flag rtcp-mux-demux<br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>> Regards,<br></div><div dir="ltr">>>>>>>> Nikita Stashkov<br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>>> 22 нояб. 2014 г., в 20:28, H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>><br></div><div dir="ltr">>>>>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>>><br></div><div dir="ltr">> написал(а):<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> Hi,<br></div><div dir="ltr">>>>>>>>> I checked you site. it seems that is a good webRTC solution.<br></div><div dir="ltr">>>>>>>>> Can you share with us your experience to solve our problem? or<br></div><div dir="ltr">>>>>>>>> any script modifications?<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> Regards,<br></div><div dir="ltr">>>>>>>>> H.Yavari<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">> ------------------------------------------------------------------------<br></div><div dir="ltr">>>>>>>>> *From:* Nikita Stashkov <<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>><br></div><div dir="ltr">>>>>>>>> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a> <mailto:<a ymailto="mailto:snl@sipmobile.org" href="mailto:snl@sipmobile.org">snl@sipmobile.org</a>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> You can try with my site - www.sipmobile.org<br></div><div dir="ltr">>>>>>>>> <<a href="http://www.sipmobile.org/" target="_blank">http://www.sipmobile.org/</a>>.<br></div><div dir="ltr">>>>>>>>> Create accounts: <a href="https://www.sipmobile.org/register/" target="_blank">https://www.sipmobile.org/register/</a><br></div><div dir="ltr">>>>>>>>> And try to call with webRTC client and SIP.<br></div><div dir="ltr">>>>>>>>> I have modified some Kamailio SPCE scripts.<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> Regards,<br></div><div dir="ltr">>>>>>>>> Nikita Stashkov<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>>> 22 нояб. 2014 г., в 16:04, Thomas Odorfer <<a ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com">odotom@gmail.com</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com">odotom@gmail.com</a>><br></div><div dir="ltr">>>>>>>>>> <mailto:<a ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com">odotom@gmail.com</a> <mailto:<a ymailto="mailto:odotom@gmail.com" href="mailto:odotom@gmail.com">odotom@gmail.com</a>>>> написал(а):<br></div><div dir="ltr">>>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> Hi,<br></div><div dir="ltr">>>>>>>>> not sure if I understood correctly which scenario works and which<br></div><div dir="ltr">>>>>>>>> not.<br></div><div dir="ltr">>>>>>>>> So browser to soft phone is now working, but what is the meaning<br></div><div dir="ltr">>>>>>>>> of browser to client? Which client?<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> I tested myself and I have to confess that I had to do some<br></div><div dir="ltr">>>>>>>>> changes in the account configs for soft phones where I am not<br></div><div dir="ltr">>>>>>>>> happy about.<br></div><div dir="ltr">>>>>>>>> It only worked between browser-webrtc and soft phone when the<br></div><div dir="ltr">>>>>>>>> corresponding account for the soft phone - nat & media flow<br></div><div dir="ltr">>>>>>>>> control had been changed to "force avp"“ and "force rtp“ ie. no<br></div><div dir="ltr">>>>>>>>> encryption.<br></div><div dir="ltr">>>>>>>>> (I have to investigate that one - could be related to an upgrade<br></div><div dir="ltr">>>>>>>>> I had performed last week - usually srtp should also work with<br></div><div dir="ltr">>>>>>>>> softphones, within the ftp.log there was „SRTP output wanted but<br></div><div dir="ltr">>>>>>>>> no crypto suite was negotiated“).<br></div><div dir="ltr">>>>>>>>> However, after my changes the following tests had been successful:<br></div><div dir="ltr">>>>>>>>> browser webrtc to softphone (eg. jitsi, counterpath x-lite -<br></div><div dir="ltr">>>>>>>>> should be software compatible with eyebeam)<br></div><div dir="ltr">>>>>>>>> browser webrtc to another browser webrtc (jssip-0.50)<br></div><div dir="ltr">>>>>>>>> browser webrtc to pstn via sip trunking (standard sip trunk,<br></div><div dir="ltr">>>>>>>>> peer settings for media force „rtp“, „force rtp“, „always with<br></div><div dir="ltr">>>>>>>>> plain SDP“)<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> That is based on the latest SPCE version 3.6.1.<br></div><div dir="ltr">>>>>>>>> What does not seem to be achievable at the moment that you can<br></div><div dir="ltr">>>>>>>>> have an account that supports „standard“ and webrtc<br></div><div dir="ltr">>>>>>>>> simultaneously ( at least I haven’t succeeded with such a setup,<br></div><div dir="ltr">>>>>>>>> maybe some sipwise/kamailio/rtpengine expert knows the trick).<br></div><div dir="ltr">>>>>>>>> And I do not have a solution yet how to share one phone number<br></div><div dir="ltr">>>>>>>>> between two accounts with different profiles.<br></div><div dir="ltr">>>>>>>>> The only solution I have at the moment is that I put a webrtc<br></div><div dir="ltr">>>>>>>>> gateway (similar to webrtc2sip from doubango) in front of SPCE<br></div><div dir="ltr">>>>>>>>> for webrtc clients.<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> For your particular problem, maybe you have to check whether your<br></div><div dir="ltr">>>>>>>>> domain settings allow „bypass rtp proxy“ behind the same NAT -<br></div><div dir="ltr">>>>>>>>> assuming you are testing wthin your LAN - this should be set to<br></div><div dir="ltr">>>>>>>>> never.<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> Good luck<br></div><div dir="ltr">>>>>>>>> Thomas<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> Am 22.11.2014 um 12:49 schrieb H Yavari <<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>><br></div><div dir="ltr">>>>>>>>> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a> <mailto:<a ymailto="mailto:hyavari@rocketmail.com" href="mailto:hyavari@rocketmail.com">hyavari@rocketmail.com</a>>>>:<br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>>> Hi,<br></div><div dir="ltr">>>>>>>>>><br></div><div dir="ltr">>>>>>>>>> I did this configs:<br></div><div dir="ltr">>>>>>>>>> use_rtpproxy: „Always with rtpptoxy as only ICE candidate“<br></div><div dir="ltr">>>>>>>>>> rtcp_feedback: „Force AVP“<br></div><div dir="ltr">>>>>>>>>> srtp_transcoding: „Force RTP“<br></div><div dir="ltr">>>>>>>>>><br></div><div dir="ltr">>>>>>>>>> now calls between browser to soft phone is ok, but browser to<br></div><div dir="ltr">>>>>>>>>> client and browser to browser receive this error "Failed to get<br></div><div dir="ltr">>>>>>>>>> local SDP"<br></div><div dir="ltr">>>>>>>>>> and calls not be established. Have you any idea about this<br></div><div dir="ltr">>>>>>>>>> situation?<br></div><div dir="ltr">>>>>>>>>> Thanks for helps.<br></div><div dir="ltr">>>>>>>>>><br></div><div dir="ltr">>>>>>>>>> Regards,<br></div><div dir="ltr">>>>>>>>>> H.Yavari<br></div><div dir="ltr">>>>>>>>>><br></div><div dir="ltr">> ------------------------------------------------------------------------<br></div><div dir="ltr">>>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>> _______________________________________________<br></div><div dir="ltr">>>>>>>>> Spce-user mailing list<br></div><div dir="ltr">>>>>>>>> <a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a>><br></div><div dir="ltr">> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a>>><br></div><div dir="ltr">>>>>>>>> <a href="https://lists.sipwise.com/listinfo/spce-user" target="_blank">https://lists.sipwise.com/listinfo/spce-user</a><br></div><div dir="ltr">> <br></div><div dir="ltr">> <br></div><div dir="ltr">> <br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>>><br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>>><br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>>><br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>>><br></div><div dir="ltr">>><br></div><div dir="ltr">>><br></div><div dir="ltr">>><br></div><div dir="ltr">>><br></div><div dir="ltr">>><br></div><div dir="ltr">>><br></div><div dir="ltr">>><br></div><div dir="ltr">>> _______________________________________________<br></div><div dir="ltr">>> Spce-user mailing list<br></div><div dir="ltr">>> <a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a>><br></div><div dir="ltr">>> <a href="https://lists.sipwise.com/listinfo/spce-user" target="_blank">https://lists.sipwise.com/listinfo/spce-user</a><br></div><div dir="ltr">>><br></div><div dir="ltr">> _______________________________________________<br></div><div dir="ltr">> Spce-user mailing list<br></div><div dir="ltr">> <a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a> <mailto:<a ymailto="mailto:Spce-user@lists.sipwise.com" href="mailto:Spce-user@lists.sipwise.com">Spce-user@lists.sipwise.com</a>><br></div><div dir="ltr">> <a href="https://lists.sipwise.com/listinfo/spce-user" target="_blank">https://lists.sipwise.com/listinfo/spce-user</a><br></div><div dir="ltr">> <br></div><div dir="ltr">> </div><br><br></div> </div> </div> </div></body></html>