<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:13px"><div id="yui_3_16_0_1_1416203358874_104815" dir="ltr"><span>Hi,</span></div><div id="yui_3_16_0_1_1416203358874_104813" dir="ltr"><span id="yui_3_16_0_1_1416203358874_104812">Thanks. I should any config for DTLS keys?</span></div><div id="yui_3_16_0_1_1416203358874_104804" dir="ltr"><br></div><div id="yui_3_16_0_1_1416203358874_104807" dir="ltr"><br><span></span></div><div id="yui_3_16_0_1_1416203358874_104802" dir="ltr"><span>Regards,</span></div><div id="yui_3_16_0_1_1416203358874_104800" dir="ltr"><span>H. Yavari</span></div><br> <div id="yui_3_16_0_1_1416203358874_104554" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;"> <div id="yui_3_16_0_1_1416203358874_104553" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div id="yui_3_16_0_1_1416203358874_104797" dir="ltr"> <hr size="1"> <font id="yui_3_16_0_1_1416203358874_104796" face="Arial" size="2"> <b><span style="font-weight:bold;">From:</span></b> Andreas Granig <agranig@sipwise.com><br> <b><span style="font-weight: bold;">To:</span></b> spce-user@lists.sipwise.com <br> <b><span style="font-weight: bold;">Sent:</span></b> Monday, 17 November 2014, 13:31:35<br> <b><span style="font-weight: bold;">Subject:</span></b> Re: [Spce-user] Fw: webRTC in production<br> </font> </div> <div id="yui_3_16_0_1_1416203358874_104552" class="y_msg_container"><br><div id="yui_3_16_0_1_1416203358874_104551" dir="ltr">Hi,<br clear="none"><br clear="none">From a general point of view, you can just point your WebRTC clients to<br clear="none">https://<ip>/wss/sip/, which is internally redirected to kamailio-lb at<br clear="none">https://<ip>:5061/ws. That's it from a signalling point of view.<br clear="none"><br clear="none">In regards to media, the SPCE can be configured for example in your<br clear="none">peering server and domain preferences to force plain RTP for your SIP<br clear="none">interconnects for PSTN origination and termination, most notably the<br clear="none">srtp_transcoding and rtcp_feedback preferences. That way, the rtpengine<br clear="none">can pick up DTLS-SRTP on your local subscriber side and transform it to<br clear="none">plain RTP on the peering side and vice versa.<br clear="none"><br clear="none">Andreas<br clear="none"><br clear="none">On 11/17/2014 10:36 AM, H Yavari wrote:<br clear="none">> <br clear="none">> Hi,<br clear="none">> Thanks for your reply. Are there any Docs and etc. for implementation?<br clear="none">> <br clear="none">> <br clear="none">> Regards,<br clear="none">> H. Yavari<br clear="none">> <br clear="none">>> ------------------------------------------------------------------------<br clear="none">>><br clear="none">>> Hi,<br clear="none">>><br clear="none">>> Within the Sipwise platforms, we're using kamailio for<br clear="none">>> SIP-over-Websocket to do the signalling and rtpengine for the SRTP-DTLS<br clear="none">>> transcoding, which is considered a highly performing and robust combo<br clear="none">>> being in use by really big Telcos in production today.<br clear="none">>><br clear="none">>> Andreas<br clear="none">>><br clear="none">>> On 11/17/2014 09:33 AM, H Yavari wrote:<br clear="none">>>> Hi,<br clear="none">>>><br clear="none">>>> I read some posts like TV over webRTC in the Sipwise website. I'm so<br clear="none">>>> interested to implement webRTC with this. want to know that this is an<br clear="none">>>> robust solution and it can be used in production? because the Asterisk<br clear="none">>>> and FS have problem with SRTP-DTLS and not support webRTC fully yet.<br clear="none">>>><br clear="none">>>> So any one have experience with webRTC implementation with Sipwise in<br clear="none">>>> production?<br clear="none">>>><br clear="none">>>><br clear="none">>>> Regards,<br clear="none">>>> H. 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