<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_1429071899625_2576"><span>Hi,</span></div><div id="yui_3_16_0_1_1429071899625_2745" dir="ltr"><br><span></span></div><div id="yui_3_16_0_1_1429071899625_2781" dir="ltr"><span id="yui_3_16_0_1_1429071899625_2975">I can configure RTP setting for peers but this is not work for me. For example I have an asterisk as a peer and I selected RTP/AVP. but SDP in calls from webrtc clients to peer is : UDP/TLS/RTP/SAVP. So asterisk reject the calls.</span></div><div id="yui_3_16_0_1_1429071899625_3078" dir="ltr"><br><span id="yui_3_16_0_1_1429071899625_2975"></span></div><div id="yui_3_16_0_1_1429071899625_3079" dir="ltr"><br><span id="yui_3_16_0_1_1429071899625_2975"></span></div><div id="yui_3_16_0_1_1429071899625_3312" dir="ltr"><span id="yui_3_16_0_1_1429071899625_2975">But about dynamically: <br></span></div><div id="yui_3_16_0_1_1429071899625_3247" dir="ltr"><span id="yui_3_16_0_1_1429071899625_2975">why I can't make a call between sip extensions and webrtc clients that are registered on ngcp? I found in log that SDP for callee doesn't change automatically.</span></div><div id="yui_3_16_0_1_1429071899625_3588" dir="ltr"><span id="yui_3_16_0_1_1429071899625_2975"><br></span></div><div id="yui_3_16_0_1_1429071899625_3589" dir="ltr"><br></div><div id="yui_3_16_0_1_1429071899625_3553" dir="ltr">Regards,</div><div id="yui_3_16_0_1_1429071899625_3623" dir="ltr">H.Yavari<br> </div><div id="yui_3_16_0_1_1429071899625_2563" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 13px;"> <div id="yui_3_16_0_1_1429071899625_2562" style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, Sans-Serif; font-size: 16px;"> <div id="yui_3_16_0_1_1429071899625_2573" dir="ltr"> <hr id="yui_3_16_0_1_1429071899625_2782" size="1"> <font id="yui_3_16_0_1_1429071899625_2572" face="Arial" size="2"> <b><span style="font-weight:bold;">From:</span></b> Andrew Pogrebennyk <apogrebennyk@sipwise.com><br> <br> </font> </div> <div id="yui_3_16_0_1_1429071899625_2561" class="y_msg_container"><br><div id="yui_3_16_0_1_1429071899625_2560" dir="ltr">On 04/14/2015 12:55 PM, H Yavari wrote:<br clear="none">> I have some problem with new version (3.8,1).<br clear="none">> webrtc clients to webrtc clients calls are ok when clients are same. I<br clear="none">> mean when caller and callee are on SIPmpl5 or Sip.js calls are ok.<br clear="none">> I think the reason is: in this version, we can't select more than one<br clear="none">> policy for RTP for a domain. So webrtc client <---> sip client calls are<br clear="none">> not possible too.<br clear="none">> <br clear="none">> In 3.7 version, we could force RTP/AVP for peer servers. but in 3.8.1<br clear="none">> this part is changed and we can't do this. So WebRTC calls to SIP legacy<br clear="none">> peers are not possible (SDP incompatible) .<br clear="none"><br clear="none">I'm not sure if I understand you, can't you set the transport protocol<br clear="none">on the peer in the mr3.8.1? The only difference between two releases,<br clear="none">from my point of view is that in mr3.8.1 you have one preference instead<br clear="none">of two (rtcp_feedback has been combined with rtp transcoding).<br clear="none"><br clear="none">> It is possible that NGCP make decision about SDP dynamically ?<br clear="none"><br clear="none">We kind of do this already, because if callee is on the websocket we<br clear="none">default to UDP/TLS/RTP/SAVPF.<div class="qtdSeparateBR"><br><br></div><div class="yqt6184514180" id="yqtfd30246"><br clear="none"><br clear="none">Andrew<br clear="none"></div></div><br><br></div> </div> </div> </div></body></html>