<div dir="ltr">Even when calling the echo call service, it does not work. To be more specific: How can I make FS respond to this SDP<div><br></div><div><div><div>v=0</div><div>o=- 7827925660507220020 2 IN IP4 127.0.0.1</div><div>s=-</div><div>t=0 0</div><div>a=group:BUNDLE audio</div><div>a=msid-semantic: WMS ARDAMS</div><div>m=audio 9 RTP/SAVPF 111 103 9 102 0 8 106 105 13 127 126</div><div>c=IN IP4 0.0.0.0</div><div>a=rtpmap:111 opus/48000/2</div><div>a=fmtp:111 minptime=10; useinbandfec=1</div><div>a=rtpmap:103 ISAC/16000</div><div>a=rtpmap:9 G722/8000</div><div>a=rtpmap:102 ILBC/8000</div><div>a=rtpmap:0 PCMU/8000</div><div>a=rtpmap:8 PCMA/8000</div><div>a=rtpmap:106 CN/32000</div><div>a=rtpmap:105 CN/16000</div><div>a=rtpmap:13 CN/8000</div><div>a=rtpmap:127 red/8000</div><div>a=rtpmap:126 telephone-event/8000</div><div>a=rtcp:9 IN IP4 0.0.0.0</div><div>a=ice-ufrag:MDEFXaTwID0Qv/el</div><div>a=ice-pwd:oj31a5bRhExyyehlEKTAVFw1</div><div>a=mid:audio</div><div>a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level</div><div>a=extmap:3 <a href="http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time">http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time</a></div><div>a=rtcp-mux</div><div>a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:iMD5gSrO/mnMCNTcp3k85tMS3P4NgXf6wFubAmZW</div><div>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:iMD5gSrO/mnMCNTcp3k85tMS3P4NgXf6wFubAmZW</div><div>a=maxptime:60</div><div>a=ssrc:2184339687 cname:/TpRcJ+yclk4xBm4</div><div>a=ssrc:2184339687 msid:ARDAMS ARDAMSa0</div><div>a=ssrc:2184339687 mslabel:ARDAMS</div><div>a=ssrc:2184339687 label:ARDAMSa0</div><div>a=candidate:4203996854 1 udp 2122260223 10.110.44.92 46241 typ host generation 0</div></div></div><div><br></div><div>with something compatible, i.e. something having ice-ufrag/ice-pwd/candidate, and crypto entries?</div><div><br></div><div>Instead, it responds with this incomplete SDP:</div><div><br></div><div><div><div>v=0</div><div>o=FreeSWITCH 1444208399 1444208400 IN IP4 10.110.36.194</div><div>s=FreeSWITCH</div><div>c=IN IP4 10.110.36.194</div><div>t=0 0</div><div>a=sendonly</div><div>a=msid-semantic: WMS hotZDrIQsyNJstePWz1e1QRgqVxi8lkg</div><div>m=audio 19770 RTP/AVPF 111 126</div></div></div><div><br></div><div>Some additional lines are printed on the FS log, but they don't seem to make it on the network (according to WireShark)</div><div><br></div><div><div>a=rtpmap:111 opus/48000/2</div><div>a=fmtp:111 useinbandfec=1; minptime=10</div><div>a=rtpmap:126 telephone-event/8000</div><div>a=ptime:20</div><div>a=rtcp-mux</div><div>a=rtcp:19770 IN IP4 10.110.36.194</div><div>a=ice-ufrag:hSw9zdfPZDYYO94d</div><div>a=ice-pwd:D05puFtnS7zoQqGY83Yx73dW</div><div>a=candidate:5758928345 1 udp 659136 10.110.36.194 19770 typ host generation 0</div><div>a=ssrc:2316904761 cname:zE52VDJe2mOlFS0s</div><div>a=ssrc:2316904761 msid:hotZDrIQsyNJstePWz1e1QRgqVxi8lkg a0</div><div>a=ssrc:2316904761 mslabel:hotZDrIQsyNJstePWz1e1QRgqVxi8lkg</div><div>a=ssrc:2316904761 label:hotZDrIQsyNJstePWz1e1QRgqVxi8lkga0</div></div><div><br></div><div>Total packet length is 1138 bytes...</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Oct 5, 2015 at 4:51 PM, Johannes Singler <span dir="ltr"><<a href="mailto:johannes.singler@qnective.com" target="_blank">johannes.singler@qnective.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">FS is a B2BUA. However, the two legs of a regular call seem to influence each other, e.g.<div><br><div>1. The caller offers an SDES-SRTP-encrypted connection with ICE.</div><div>2. FS offers to the callee a simple RTP connection (no SRTP, no ICE, as configured in the dialplan), callee answers respectively.</div><div>3. FS answers caller, but without with neither "a:crypto" entries nor ICE candidates.</div><div>4. Why is that? Shouldn't it answer SDES-SRTP with ICE to the original caller, respecting the original caller's offer?</div><div><br></div><div>When doing WebRTC from the caller (DTLS-SRTP with ICE), this actually works fine (callee unchanged).</div><div><br></div><div>So what's the general scheme for choosing encryption on either side?</div><div><br></div><div>Related to that:</div><div>Can you enable ICE without completely enabling WebRTC (media_webrtc=true) from the dialplan? That would help maybe...</div><span class="HOEnZb"><font color="#888888"><div><br></div><div>-- <br><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><span style="font-size:12.7273px">Johannes Singler</span><br style="font-size:12.7273px"><span style="font-size:12.7273px">Software Engineer</span><br style="font-size:12.7273px"><div><span style="font-size:12.7273px"><br></span></div><div><span style="font-size:12.7273px">Qnective</span><br></div></div></div></div></div></div>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><span style="font-size:12.7272720336914px">Johannes Singler</span><br style="font-size:12.7272720336914px"><span style="font-size:12.7272720336914px">Software Engineer</span><br style="font-size:12.7272720336914px"><div><span style="font-size:12.7272720336914px"><br></span></div><div><span style="font-size:12.7272720336914px">Qnective</span><br style="font-size:12.7272720336914px"><br style="font-size:12.7272720336914px"><span style="font-size:12.7272720336914px">Thurgauerstrasse 54 | 8050 Zürich | Switzerland</span><br style="font-size:12.7272720336914px"><span style="font-size:12.7272720336914px">Mobile</span> +41798379869<br style="font-size:12.7272720336914px"><a href="http://www.qnective.com/" style="color:rgb(17,85,204);font-size:12.7272720336914px" target="_blank">www.qnective.com</a><span style="font-size:12.7272720336914px"> | </span><a href="mailto:johannes.singler@qnective.com" style="color:rgb(17,85,204);font-size:12.7272720336914px" target="_blank">johannes.singler@qnective.com</a></div></div></div></div></div></div></div></div></div>
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