[asterisk-users] WebRTC / Rejecting secure audio stream errors

[Daniel Pocock]

I've seen the following appear in some tests with Asterisk 11.11:

 WARNING[3938][C-00000003]: chan_sip.c:10535 process_sdp: Rejecting
secure audio stream without encryption details: audio 54908
UDP/TLS/RTP/SAVPF 109 0 8 101


Specifically, it always happens from a Firefox 24 host but it works
without this error from another host running Firefox 26

I did a diff on the SDP and couldn't see anything obviously different
except one thing: Firefox 24 only has host candidates for ICE (TURN
support was only added in Firefox 25).  Is there any way that could
cause this error though?  It appears the encryption details are
sufficient and do not otherwise differ between Firefox 24 and 26:

--- ff-24.txt   2014-08-25 15:02:20.452383599 +0200
+++ ff-26.txt   2014-08-25 15:01:42.472346613 +0200
@@ -1,12 +1,12 @@
 v=0
-o=Mozilla-SIPUA-24.7.0 14737 0 IN IP4 0.0.0.0
+o=Mozilla-SIPUA-26.0 18111 0 IN IP4 0.0.0.0
 s=SIP Call
 t=0 0
-a=ice-ufrag:301212e4
-a=ice-pwd:d7430f468514f1f2d326d3c944691fbf
-a=fingerprint:sha-256
E2:53:6A:FA:6D:E2:3F:7E:24:82:0F:E3:27:34:D1:CC:50:31:42:82:5F:DF:34:9A:4F:42:D1:6D:B7:DB:5C:43
-m=audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101
-c=IN IP4 10.10.1.144
+a=ice-ufrag:2ff98ac6
+a=ice-pwd:dc22648d73c4b421274f31c1953828d4
+a=fingerprint:sha-256
F7:52:A3:46:A4:C3:99:36:83:05:7A:8F:B6:CC:A9:17:0A:45:04:79:3D:D7:F5:39:BE:1D:F3:FF:DA:81:DB:7C
+m=audio 51390 UDP/TLS/RTP/SAVPF 109 0 8 101
+c=IN IP4 195.8.117.59
 a=rtpmap:109 opus/48000/2
 a=ptime:20
 a=rtpmap:0 PCMU/8000
@@ -14,17 +14,21 @@
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15
 a=sendrecv
-a=candidate:0 1 UDP 2113667327 10.10.1.144 54908 typ host
-a=candidate:1 1 UDP 2113667327 192.168.1.161 52081 typ host
-a=candidate:2 1 UDP 2113667327 195.8.117.161 54978 typ host
-a=candidate:0 2 UDP 2113667326 10.10.1.144 58499 typ host
-a=candidate:1 2 UDP 2113667326 192.168.1.161 33161 typ host
-a=candidate:2 2 UDP 2113667326 195.8.117.161 36491 typ host
+a=setup:actpass
+a=candidate:0 1 UDP 2122252543 10.10.1.90 60221 typ host
+a=candidate:1 1 UDP 1686110207 195.8.117.200 60221 typ srflx raddr
10.10.1.90 rport 60221
+a=candidate:2 1 UDP 8388607 195.8.117.59 51390 typ relay raddr
195.8.117.59 rport 51390
+a=candidate:3 1 UDP 2122187007 192.168.150.1 38505 typ host
+a=candidate:0 2 UDP 2122252542 10.10.1.90 55368 typ host
+a=candidate:1 2 UDP 1686110206 195.8.117.200 55368 typ srflx raddr
10.10.1.90 rport 55368
+a=candidate:2 2 UDP 8388606 195.8.117.59 51391 typ relay raddr
195.8.117.59 rport 51391
+a=candidate:3 2 UDP 2122187006 192.168.150.1 46478 typ host
+a=rtcp-mux
 <------------->
---- (22 headers 22 lines) ---
+--- (22 headers 26 lines) ---
 Sending to 195.8.117.60:5060 (no NAT)
 Sending to 195.8.117.60:5060 (no NAT)
-Using INVITE request as basis request - kbr110264479udsqistu
+Using INVITE request as basis request - hqs8q0vi6pgckcu59a8r
 Found peer 'example.org' for 'anonymous' from 195.8.117.60:5060
   == Using SIP RTP CoS mark 5
 Found RTP audio format 109
@@ -35,5 +39,53 @@
 Found audio description format PCMU for ID 0
 Found audio description format PCMA for ID 8
 Found audio description format telephone-event for ID 101
-[Aug 25 14:59:29] WARNING[3938][C-00000003]: chan_sip.c:10535
process_sdp: Rejecting secure audio stream without encryption details:
audio 54908 UDP/TLS/RTP/SAVPF 109 0 8 101
+Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer -
audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
+Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
+Peer audio RTP is at port 195.8.117.59:51390