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Asterisk SIP Debug output with WebRTC

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  1. Asterisk 13.16.0, Copyright (C) 1999 - 2014, Digium, Inc. and others.
  2. Created by Mark Spencer <markster@digium.com>
  3. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  4. This is free software, with components licensed under the GNU General Public
  5. License version 2 and other licenses; you are welcome to redistribute it under
  6. certain conditions. Type 'core show license' for details.
  7. =========================================================================
  8. Connected to Asterisk 13.16.0 currently running on osboxes (pid = 2457)
  9. osboxes*CLI>
  10. 
  11. <--- SIP read from WS:192.168.1.75:43892 --->
  12. INVITE sip:6002@192.168.1.91 SIP/2.0
  13. Via: SIP/2.0/WS 192.0.2.240;branch=z9hG4bK4047956
  14. Max-Forwards: 70
  15. To: <sip:6002@192.168.1.91>
  16. From: <sip:demo-alice@192.168.1.91>;tag=6ugdi4fseq
  17. Call-ID: m697iknsn33aa0et75a5
  18. CSeq: 1630 INVITE
  19. Contact: <sip:n5el0164@192.0.2.240;transport=ws;ob>
  20. Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE
  21. Content-Type: application/sdp
  22. Supported: outbound
  23. User-Agent: SIP.js/0.7.2
  24. Content-Length: 1470
  25.  
  26. v=0
  27. o=- 5737134974404490177 2 IN IP4 127.0.0.1
  28. s=-
  29. t=0 0
  30. a=group:BUNDLE audio
  31. a=msid-semantic: WMS 0lLKetBudMpPaoylXOwhqy67MaAdEJK2Jkn1
  32. m=audio 42029 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
  33. c=IN IP4 88.75.105.210
  34. a=rtcp:9 IN IP4 192.0.2.137
  35. a=candidate:1268038522 1 udp 2113937151 192.168.1.75 42029 typ host generation 0 network-cost 50
  36. a=candidate:842163049 1 udp 1677729535 88.75.105.210 42029 typ srflx raddr 192.168.1.75 rport 42029 generation 0 network-cost 50
  37. a=ice-ufrag:qEiO
  38. a=ice-pwd:TRDqiEBGdtj8yBa2xbyjnVmY
  39. a=fingerprint:sha-256 22:D3:F3:F8:FF:36:D7:C1:09:AD:65:16:85:00:1C:80:0C:47:F9:5A:7B:18:D9:F5:F6:7D:88:85:22:E8:89:74
  40. a=setup:actpass
  41. a=mid:audio
  42. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  43. a=sendrecv
  44. a=rtcp-mux
  45. a=rtpmap:111 opus/48000/2
  46. a=rtcp-fb:111 transport-cc
  47. a=fmtp:111 minptime=10;useinbandfec=1
  48. a=rtpmap:103 ISAC/16000
  49. a=rtpmap:104 ISAC/32000
  50. a=rtpmap:9 G722/8000
  51. a=rtpmap:0 PCMU/8000
  52. a=rtpmap:8 PCMA/8000
  53. a=rtpmap:106 CN/32000
  54. a=rtpmap:105 CN/16000
  55. a=rtpmap:13 CN/8000
  56. a=rtpmap:110 telephone-event/48000
  57. a=rtpmap:112 telephone-event/32000
  58. a=rtpmap:113 telephone-event/16000
  59. a=rtpmap:126 telephone-event/8000
  60. a=ssrc:3607560285 cname:13j7GcFfXqZwk5fS
  61. a=ssrc:3607560285 msid:0lLKetBudMpPaoylXOwhqy67MaAdEJK2Jkn1 22341503-dbeb-4e45-9f9e-1d84edda0bab
  62. a=ssrc:3607560285 mslabel:0lLKetBudMpPaoylXOwhqy67MaAdEJK2Jkn1
  63. a=ssrc:3607560285 label:22341503-dbeb-4e45-9f9e-1d84edda0bab
  64. <------------->
  65. --- (13 headers 38 lines) ---
  66. Using INVITE request as basis request - m697iknsn33aa0et75a5
  67. Found peer 'demo-alice' for 'demo-alice' from 192.168.1.75:43892
  68.  
  69. <--- Reliably Transmitting (no NAT) to 192.168.1.75:5060 --->
  70. SIP/2.0 401 Unauthorized
  71. Via: SIP/2.0/WS 192.0.2.240;branch=z9hG4bK4047956;received=192.168.1.75
  72. From: <sip:demo-alice@192.168.1.91>;tag=6ugdi4fseq
  73. To: <sip:6002@192.168.1.91>;tag=as3943adab
  74. Call-ID: m697iknsn33aa0et75a5
  75. CSeq: 1630 INVITE
  76. Server: Asterisk PBX 13.16.0
  77. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  78. Supported: replaces, timer
  79. WWW-Authenticate: Digest algorithm=MD5, realm="192.168.1.91", nonce="7923b919"
  80. Content-Length: 0
  81.  
  82.  
  83. <------------>
  84. Scheduling destruction of SIP dialog 'm697iknsn33aa0et75a5' in 32000 ms (Method: INVITE)
  85.  
  86. osboxes*CLI>
  87. 
  88. <--- SIP read from WS:192.168.1.75:43892 --->
  89. ACK sip:6002@192.168.1.91 SIP/2.0
  90. Via: SIP/2.0/WS 192.0.2.240;branch=z9hG4bK4047956
  91. To: <sip:6002@192.168.1.91>;tag=as3943adab
  92. From: <sip:demo-alice@192.168.1.91>;tag=6ugdi4fseq
  93. Call-ID: m697iknsn33aa0et75a5
  94. Content-Length: 0
  95. CSeq: 1630 ACK
  96.  
  97. <------------->
  98. --- (7 headers 0 lines) ---
  99.  
  100. <--- SIP read from WS:192.168.1.75:43892 --->
  101. INVITE sip:6002@192.168.1.91 SIP/2.0
  102. Via: SIP/2.0/WS 192.0.2.240;branch=z9hG4bK6318963
  103. Max-Forwards: 70
  104. To: <sip:6002@192.168.1.91>
  105. From: <sip:demo-alice@192.168.1.91>;tag=6ugdi4fseq
  106. Call-ID: m697iknsn33aa0et75a5
  107. CSeq: 1631 INVITE
  108. Authorization: Digest algorithm=MD5, username="demo-alice", realm="192.168.1.91", nonce="7923b919", uri="sip:6002@192.168.1.91", response="0d756105c37a26e6bc1484f9fec47075"
  109. Contact: <sip:n5el0164@192.0.2.240;transport=ws;ob>
  110. Allow: ACK,CANCEL,BYE,OPTIONS,INFO,NOTIFY,INVITE
  111. Content-Type: application/sdp
  112. Supported: outbound
  113. User-Agent: SIP.js/0.7.2
  114. Content-Length: 1470
  115.  
  116. v=0
  117. o=- 5737134974404490177 2 IN IP4 127.0.0.1
  118. s=-
  119. t=0 0
  120. a=group:BUNDLE audio
  121. a=msid-semantic: WMS 0lLKetBudMpPaoylXOwhqy67MaAdEJK2Jkn1
  122. m=audio 42029 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
  123. c=IN IP4 88.75.105.210
  124. a=rtcp:9 IN IP4 192.0.2.137
  125. a=candidate:1268038522 1 udp 2113937151 192.168.1.75 42029 typ host generation 0 network-cost 50
  126. a=candidate:842163049 1 udp 1677729535 88.75.105.210 42029 typ srflx raddr 192.168.1.75 rport 42029 generation 0 network-cost 50
  127. a=ice-ufrag:qEiO
  128. a=ice-pwd:TRDqiEBGdtj8yBa2xbyjnVmY
  129. a=fingerprint:sha-256 22:D3:F3:F8:FF:36:D7:C1:09:AD:65:16:85:00:1C:80:0C:47:F9:5A:7B:18:D9:F5:F6:7D:88:85:22:E8:89:74
  130. a=setup:actpass
  131. a=mid:audio
  132. a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
  133. a=sendrecv
  134. a=rtcp-mux
  135. a=rtpmap:111 opus/48000/2
  136. a=rtcp-fb:111 transport-cc
  137. a=fmtp:111 minptime=10;useinbandfec=1
  138. a=rtpmap:103 ISAC/16000
  139. a=rtpmap:104 ISAC/32000
  140. a=rtpmap:9 G722/8000
  141. a=rtpmap:0 PCMU/8000
  142. a=rtpmap:8 PCMA/8000
  143. a=rtpmap:106 CN/32000
  144. a=rtpmap:105 CN/16000
  145. a=rtpmap:13 CN/8000
  146. a=rtpmap:110 telephone-event/48000
  147. a=rtpmap:112 telephone-event/32000
  148. a=rtpmap:113 telephone-event/16000
  149. a=rtpmap:126 telephone-event/8000
  150. a=ssrc:3607560285 cname:13j7GcFfXqZwk5fS
  151. a=ssrc:3607560285 msid:0lLKetBudMpPaoylXOwhqy67MaAdEJK2Jkn1 22341503-dbeb-4e45-9f9e-1d84edda0bab
  152. a=ssrc:3607560285 mslabel:0lLKetBudMpPaoylXOwhqy67MaAdEJK2Jkn1
  153. a=ssrc:3607560285 label:22341503-dbeb-4e45-9f9e-1d84edda0bab
  154. <------------->
  155. --- (14 headers 38 lines) ---
  156. Using INVITE request as basis request - m697iknsn33aa0et75a5
  157. Found peer 'demo-alice' for 'demo-alice' from 192.168.1.75:43892
  158. [Jun 30 14:47:29] NOTICE[2800][C-0000000f]: chan_sip.c:10408 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 42029 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
  159. Found RTP audio format 111
  160. Found RTP audio format 103
  161. Found RTP audio format 104
  162. Found RTP audio format 9
  163. Found RTP audio format 0
  164. Found RTP audio format 8
  165. Found RTP audio format 106
  166. Found RTP audio format 105
  167. Found RTP audio format 13
  168. Found RTP audio format 110
  169. Found RTP audio format 112
  170. Found RTP audio format 113
  171. Found RTP audio format 126
  172. Found audio description format opus for ID 111
  173. Found unknown media description format ISAC for ID 103
  174. Found unknown media description format ISAC for ID 104
  175. Found audio description format G722 for ID 9
  176. Found audio description format PCMU for ID 0
  177. Found audio description format PCMA for ID 8
  178. Found unknown media description format CN for ID 106
  179. Found unknown media description format CN for ID 105
  180. Found audio description format CN for ID 13
  181. Found unknown media description format telephone-event for ID 110
  182. Found unknown media description format telephone-event for ID 112
  183. Found unknown media description format telephone-event for ID 113
  184. Found audio description format telephone-event for ID 126
  185. [Jun 30 14:47:29] WARNING[2800][C-0000000f]: chan_sip.c:10807 process_sdp: Rejecting secure audio stream without encryption details: audio 42029 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
  186.  
  187. <--- Reliably Transmitting (no NAT) to 192.168.1.75:5060 --->
  188. SIP/2.0 488 Not acceptable here
  189. Via: SIP/2.0/WS 192.0.2.240;branch=z9hG4bK6318963;received=192.168.1.75
  190. From: <sip:demo-alice@192.168.1.91>;tag=6ugdi4fseq
  191. To: <sip:6002@192.168.1.91>;tag=as3943adab
  192. Call-ID: m697iknsn33aa0et75a5
  193. CSeq: 1631 INVITE
  194. Server: Asterisk PBX 13.16.0
  195. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  196. Supported: replaces, timer
  197. Content-Length: 0
  198.  
  199.  
  200. <------------>
  201. Scheduling destruction of SIP dialog 'm697iknsn33aa0et75a5' in 32000 ms (Method: INVITE)
  202.  
  203. osboxes*CLI>
  204. 
  205. <--- SIP read from WS:192.168.1.75:43892 --->
  206. ACK sip:6002@192.168.1.91 SIP/2.0
  207. Via: SIP/2.0/WS 192.0.2.240;branch=z9hG4bK6318963
  208. To: <sip:6002@192.168.1.91>;tag=as3943adab
  209. From: <sip:demo-alice@192.168.1.91>;tag=6ugdi4fseq
  210. Call-ID: m697iknsn33aa0et75a5
  211. Content-Length: 0
  212. CSeq: 1631 ACK
  213.  
  214. <------------->
  215. --- (7 headers 0 lines) ---
  216.  
  217. osboxes*CLI> exit
  218. Asterisk cleanly ending (0).
  219. Executing last minute cleanups
  220. 
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