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This call needs video offers, but caller probably did not...

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Mar 29th, 2013
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  1. <--- SIP read from UDP:89.209.100.91:22980 --->
  2. INVITE sip:6000444@209.239.114.51:5060 SIP/2.0
  3. Via: SIP/2.0/UDP 192.168.3.96:22980;rport;branch=z9hG4bK227466744
  4. From: <sip:6000333@vm.intersog.com:22071>;tag=1351731604
  5. To: "Patient" <sip:6000444@209.239.114.51>;tag=as7db15ebe
  6. Call-ID: 291853999
  7. CSeq: 22 INVITE
  8. Contact: <sip:linphone.iphone@89.209.100.91:22980>
  9. Content-Type: application/sdp
  10. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  11. Max-Forwards: 70
  12. User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  13. Subject: Media change
  14. Content-Length: 532
  15.  
  16. v=0
  17. o=6000333 1786 3814 IN IP4 192.168.3.96
  18. s=Talk
  19. c=IN IP4 192.168.3.96
  20. b=AS:380
  21. t=0 0
  22. m=audio 7076 RTP/AVP 120 111 110 0 8 121 9 3 101
  23. a=rtpmap:120 SILK/16000
  24. a=rtpmap:111 speex/16000
  25. a=fmtp:111 vbr=on
  26. a=rtpmap:110 speex/8000
  27. a=fmtp:110 vbr=on
  28. a=rtpmap:121 SILK/24000
  29. a=rtpmap:9 G722/8000
  30. a=rtpmap:101 telephone-event/8000
  31. a=fmtp:101 0-11
  32. m=video 9078 RTP/AVP 103 102 99
  33. a=rtpmap:103 VP8/90000
  34. a=rtpmap:102 H264/90000
  35. a=fmtp:102 profile-level-id=428014
  36. a=rtpmap:99 MP4V-ES/90000
  37. a=fmtp:99 profile-level-id=3
  38. <------------->
  39. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 0 [ 46]: INVITE sip:6000444@209.239.114.51:5060 SIP/2.0
  40. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 192.168.3.96:22980;rport;branch=z9hG4bK227466744
  41. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: <sip:6000333@vm.intersog.com:22071>;tag=1351731604
  42. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 3 [ 57]: To: "Patient" <sip:6000444@209.239.114.51>;tag=as7db15ebe
  43. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 4 [ 18]: Call-ID: 291853999
  44. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 5 [ 15]: CSeq: 22 INVITE
  45. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 6 [ 50]: Contact: <sip:linphone.iphone@89.209.100.91:22980>
  46. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp
  47. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 8 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  48. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 9 [ 16]: Max-Forwards: 70
  49. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 10 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  50. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 11 [ 21]: Subject: Media change
  51. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 12 [ 19]: Content-Length: 532
  52. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 13 [ 0]:
  53. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0
  54. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000333 1786 3814 IN IP4 192.168.3.96
  55. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk
  56. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 3 [ 21]: c=IN IP4 192.168.3.96
  57. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380
  58. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0
  59. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 6 [ 48]: m=audio 7076 RTP/AVP 120 111 110 0 8 121 9 3 101
  60. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 7 [ 23]: a=rtpmap:120 SILK/16000
  61. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 8 [ 24]: a=rtpmap:111 speex/16000
  62. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 9 [ 17]: a=fmtp:111 vbr=on
  63. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 10 [ 23]: a=rtpmap:110 speex/8000
  64. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 11 [ 17]: a=fmtp:110 vbr=on
  65. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 12 [ 23]: a=rtpmap:121 SILK/24000
  66. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 13 [ 20]: a=rtpmap:9 G722/8000
  67. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 14 [ 33]: a=rtpmap:101 telephone-event/8000
  68. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 15 [ 15]: a=fmtp:101 0-11
  69. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 16 [ 31]: m=video 9078 RTP/AVP 103 102 99
  70. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 17 [ 22]: a=rtpmap:103 VP8/90000
  71. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 18 [ 23]: a=rtpmap:102 H264/90000
  72. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 19 [ 34]: a=fmtp:102 profile-level-id=428014
  73. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 20 [ 25]: a=rtpmap:99 MP4V-ES/90000
  74. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9605 parse_request: Body 21 [ 28]: a=fmtp:99 profile-level-id=3
  75. --- (13 headers 22 lines) ---
  76. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9118 find_call: = Looking for Call ID: 291853999 (Checking From) --From tag 1351731604 --To-tag as7db15ebe
  77. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000035c] bound to thread.
  78. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:27866 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE
  79. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.96:22980' into...
  80. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.96' and port '22980'.
  81. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:17845 check_via: NAT detected for 192.168.3.96:22980 / 89.209.100.91:22980
  82. Sending to 89.209.100.91:22980 (NAT)
  83. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:24999 handle_request_invite: Initializing initreq for method INVITE - callid 291853999
  84. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
  85. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000333 1786 3814 IN IP4 192.168.3.96... OK.
  86. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED.
  87. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.96' into...
  88. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.96' and port ''.
  89. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 192.168.3.96... OK.
  90. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED.
  91. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
  92. Found RTP audio format 120
  93. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 120 based on m type on 0x7fa820070630
  94. Found RTP audio format 111
  95. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 111 based on m type on 0x7fa820070630
  96. Found RTP audio format 110
  97. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 110 based on m type on 0x7fa820070630
  98. Found RTP audio format 0
  99. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7fa820070630
  100. Found RTP audio format 8
  101. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 8 based on m type on 0x7fa820070630
  102. Found RTP audio format 121
  103. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 121 based on m type on 0x7fa820070630
  104. Found RTP audio format 9
  105. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa820070630
  106. Found RTP audio format 3
  107. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0x7fa820070630
  108. Found RTP audio format 101
  109. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa820070630
  110. Found audio description format SILK for ID 120
  111. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:120 SILK/16000... OK.
  112. Found audio description format speex for ID 111
  113. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:111 speex/16000... OK.
  114. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:111 vbr=on... OK.
  115. Found audio description format speex for ID 110
  116. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:110 speex/8000... OK.
  117. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:110 vbr=on... OK.
  118. Found audio description format SILK for ID 121
  119. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:121 SILK/24000... OK.
  120. Found audio description format G722 for ID 9
  121. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
  122. Found audio description format telephone-event for ID 101
  123. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  124. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED.
  125. Found RTP video format 103
  126. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 103 based on m type on 0x7fa820074980
  127. Found RTP video format 102
  128. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 102 based on m type on 0x7fa820074980
  129. Found RTP video format 99
  130. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 99 based on m type on 0x7fa820074980
  131. Found video description format VP8 for ID 103
  132. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=rtpmap:103 VP8/90000... OK.
  133. Found video description format H264 for ID 102
  134. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=rtpmap:102 H264/90000... OK.
  135. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=fmtp:102 profile-level-id=428014... OK.
  136. Found video description format MP4V-ES for ID 99
  137. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=rtpmap:99 MP4V-ES/90000... OK.
  138. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (video) SDP a=fmtp:99 profile-level-id=3... OK.
  139. Capabilities: us - (gsm|ulaw|g729|g722|h263|h263p|h264|mpeg4|vp8), peer - audio=(gsm|ulaw|alaw|speex|speex16|g722|silk16|silk24)/video=(h264|mpeg4|vp8)/text=(nothing), combined - (gsm|ulaw|g722|h264|mpeg4|vp8)
  140. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  141. Peer audio RTP is at port 192.168.3.96:7076
  142. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x7fa820070630 to 0x7fa7f0509a50
  143. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 3 from 0x7fa820070630 to 0x7fa7f0509a50
  144. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 8 from 0x7fa820070630 to 0x7fa7f0509a50
  145. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa820070630 to 0x7fa7f0509a50
  146. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa820070630 to 0x7fa7f0509a50
  147. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 110 from 0x7fa820070630 to 0x7fa7f0509a50
  148. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 111 from 0x7fa820070630 to 0x7fa7f0509a50
  149. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 120 from 0x7fa820070630 to 0x7fa7f0509a50
  150. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 121 from 0x7fa820070630 to 0x7fa7f0509a50
  151. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa7f0509888'
  152. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into...
  153. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''.
  154. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into...
  155. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''.
  156. Peer video RTP is at port 192.168.3.96:9078
  157. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 99 from 0x7fa820074980 to 0x7fa7f059a100
  158. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 102 from 0x7fa820074980 to 0x7fa7f059a100
  159. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 103 from 0x7fa820074980 to 0x7fa7f059a100
  160. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa7f05af668'
  161. Peer doesn't provide T.140
  162. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10668 process_sdp: We're settling with these formats: (gsm|ulaw|g722|h264|mpeg4|vp8)
  163. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call
  164. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (gsm|ulaw|g722|h264|mpeg4|vp8), old nativeformats (ulaw)
  165. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:25235 handle_request_invite: Got a SIP re-invite for call 291853999
  166. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:25511 handle_request_invite: SIP/6000333-000001b3: This call is UP....
  167.  
  168. <--- Transmitting (NAT) to 89.209.100.91:22980 --->
  169. SIP/2.0 100 Trying
  170. Via: SIP/2.0/UDP 192.168.3.96:22980;branch=z9hG4bK227466744;received=89.209.100.91;rport=22980
  171. From: <sip:6000333@vm.intersog.com:22071>;tag=1351731604
  172. To: "Patient" <sip:6000444@209.239.114.51>;tag=as7db15ebe
  173. Call-ID: 291853999
  174. CSeq: 22 INVITE
  175. Server: Video Medicine PBX
  176. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  177. Supported: replaces, timer
  178. Contact: <sip:6000444@209.239.114.51:5060>
  179. Content-Length: 0
  180.  
  181.  
  182. <------------>
  183. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 89.209.100.91:22980
  184. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:13504 transmit_response_with_sdp: Setting framing from config on incoming call
  185. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:13001 add_sdp: This call needs video offers, but caller probably did not offer it!
  186. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:13054 add_sdp: ** Our capability: (gsm|ulaw|g722|h264|mpeg4|vp8) Video flag: False Text flag: True
  187. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (nothing)
  188. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:13059 add_sdp: ** Our native-bridge filtered capablity: (gsm|ulaw|g722)
  189. Audio is at 15562
  190. Adding codec 100003 (ulaw) to SDP
  191. Adding codec 100012 (g722) to SDP
  192. Adding codec 100002 (gsm) to SDP
  193. Adding non-codec 0x1 (telephone-event) to SDP
  194. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP
  195. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (gsm|ulaw|g722)
  196.  
  197. <--- Reliably Transmitting (NAT) to 89.209.100.91:22980 --->
  198. SIP/2.0 200 OK
  199. Via: SIP/2.0/UDP 192.168.3.96:22980;branch=z9hG4bK227466744;received=89.209.100.91;rport=22980
  200. From: <sip:6000333@vm.intersog.com:22071>;tag=1351731604
  201. To: "Patient" <sip:6000444@209.239.114.51>;tag=as7db15ebe
  202. Call-ID: 291853999
  203. CSeq: 22 INVITE
  204. Server: Video Medicine PBX
  205. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  206. Supported: replaces, timer
  207. Contact: <sip:6000444@209.239.114.51:5060>
  208. Content-Type: application/sdp
  209. Content-Length: 338
  210.  
  211. v=0
  212. o=root 1759697645 1759697647 IN IP4 192.168.3.180
  213. s=Asterisk PBX 11.2.1
  214. c=IN IP4 192.168.3.180
  215. t=0 0
  216. m=audio 7076 RTP/AVP 0 9 3 101
  217. a=rtpmap:0 PCMU/8000
  218. a=rtpmap:9 G722/8000
  219. a=rtpmap:3 GSM/8000
  220. a=rtpmap:101 telephone-event/8000
  221. a=fmtp:101 0-16
  222. a=silenceSupp:off - - - -
  223. a=ptime:20
  224. a=sendrecv
  225. m=video 0 RTP/AVP 103 102 99
  226. <------------>
  227. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #93530
  228. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 89.209.100.91:22980
  229. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: res_rtp_asterisk.c:2085 ast_rtp_update_source: Setting the marker bit due to a source update
  230. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: rtp_engine.c:1278 remote_bridge_loop: Oooh, 'SIP/6000333-000001b3' changed end address to 192.168.3.96:7076 (format (gsm|ulaw|alaw|speex|speex16|g722|h264|mpeg4|vp8|silk16|silk24))
  231. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: rtp_engine.c:1281 remote_bridge_loop: Oooh, 'SIP/6000333-000001b3' was 192.168.3.96:7076/(format (gsm|ulaw|alaw|speex|speex16|g722|h264|mpeg4|vp8|silk16|silk24))
  232. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:32432 sip_set_rtp_peer: Sending reinvite on SIP '0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060' - It's audio soon redirected to IP 192.168.3.96:7076
  233. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:11802 reqprep: Strict routing enforced for session 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  234. set_destination: Parsing <sip:linphone.iphone@192.168.3.180:4481> for address/port to send to
  235. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.180:4481' into...
  236. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.180' and port '4481'.
  237. set_destination: set destination to 192.168.3.180:4481
  238. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:13054 add_sdp: ** Our capability: (gsm|ulaw|g722) Video flag: True Text flag: True
  239. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:13055 add_sdp: ** Our prefcodec: (ulaw)
  240. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:13059 add_sdp: ** Our native-bridge filtered capablity: (gsm|ulaw|g722)
  241. Audio is at 10684
  242. Adding codec 100003 (ulaw) to SDP
  243. Adding codec 100012 (g722) to SDP
  244. Adding codec 100002 (gsm) to SDP
  245. Adding non-codec 0x1 (telephone-event) to SDP
  246. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:13192 add_sdp: -- Done with adding codecs to SDP
  247. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:13390 add_sdp: Done building SDP. Settling with this capability: (gsm|ulaw|g722)
  248. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:3505 initialize_initreq: Initializing already initialized SIP dialog 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060 (presumably reinvite)
  249. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 0 [ 53]: INVITE sip:linphone.iphone@192.168.3.180:4481 SIP/2.0
  250. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 1 [ 65]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK014e74de;rport
  251. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 2 [ 16]: Max-Forwards: 70
  252. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 3 [ 58]: From: "Doctor" <sip:6000333@209.239.114.51>;tag=as4436c5b0
  253. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 4 [ 72]: To: <sip:6000444@89.209.100.91:4481;line=14bc6bcb7ab78b7>;tag=1613855871
  254. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 5 [ 42]: Contact: <sip:6000333@209.239.114.51:5060>
  255. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 6 [ 61]: Call-ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  256. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 7 [ 16]: CSeq: 105 INVITE
  257. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 8 [ 30]: User-Agent: Video Medicine PBX
  258. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 9 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  259. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 10 [ 26]: Supported: replaces, timer
  260. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 11 [ 52]: X-asterisk-Info: SIP re-invite (External RTP bridge)
  261. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:9568 parse_request: Header 12 [ 29]: Content-Type: application/sdp
  262. Reliably Transmitting (NAT) to 89.209.100.91:4481:
  263. INVITE sip:linphone.iphone@192.168.3.180:4481 SIP/2.0
  264. Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK014e74de;rport
  265. Max-Forwards: 70
  266. From: "Doctor" <sip:6000333@209.239.114.51>;tag=as4436c5b0
  267. To: <sip:6000444@89.209.100.91:4481;line=14bc6bcb7ab78b7>;tag=1613855871
  268. Contact: <sip:6000333@209.239.114.51:5060>
  269. Call-ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  270. CSeq: 105 INVITE
  271. User-Agent: Video Medicine PBX
  272. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  273. Supported: replaces, timer
  274. X-asterisk-Info: SIP re-invite (External RTP bridge)
  275. Content-Type: application/sdp
  276. Content-Length: 308
  277.  
  278. v=0
  279. o=root 1538395523 1538395526 IN IP4 192.168.3.96
  280. s=Asterisk PBX 11.2.1
  281. c=IN IP4 192.168.3.96
  282. t=0 0
  283. m=audio 7076 RTP/AVP 0 9 3 101
  284. a=rtpmap:0 PCMU/8000
  285. a=rtpmap:9 G722/8000
  286. a=rtpmap:3 GSM/8000
  287. a=rtpmap:101 telephone-event/8000
  288. a=fmtp:101 0-16
  289. a=silenceSupp:off - - - -
  290. a=ptime:20
  291. a=sendrecv
  292.  
  293. ---
  294. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:4324 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #93531
  295. [Mar 29 10:34:34] DEBUG[10929][C-0000035c]: chan_sip.c:3864 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 89.209.100.91:4481
  296. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000035c] being removed from thread.
  297. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:4108 retrans_pkt: SIP TIMER: Rescheduling retransmission #93530 (1) SIP/2.0 - 1
  298. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:4128 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 402 ms (t1 201 ms (Retrans id #93530))
  299. Retransmitting #1 (NAT) to 89.209.100.91:22980:
  300. SIP/2.0 200 OK
  301. Via: SIP/2.0/UDP 192.168.3.96:22980;branch=z9hG4bK227466744;received=89.209.100.91;rport=22980
  302. From: <sip:6000333@vm.intersog.com:22071>;tag=1351731604
  303. To: "Patient" <sip:6000444@209.239.114.51>;tag=as7db15ebe
  304. Call-ID: 291853999
  305. CSeq: 22 INVITE
  306. Server: Video Medicine PBX
  307. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  308. Supported: replaces, timer
  309. Contact: <sip:6000444@209.239.114.51:5060>
  310. Content-Type: application/sdp
  311. Content-Length: 338
  312.  
  313. v=0
  314. o=root 1759697645 1759697647 IN IP4 192.168.3.180
  315. s=Asterisk PBX 11.2.1
  316. c=IN IP4 192.168.3.180
  317. t=0 0
  318. m=audio 7076 RTP/AVP 0 9 3 101
  319. a=rtpmap:0 PCMU/8000
  320. a=rtpmap:9 G722/8000
  321. a=rtpmap:3 GSM/8000
  322. a=rtpmap:101 telephone-event/8000
  323. a=fmtp:101 0-16
  324. a=silenceSupp:off - - - -
  325. a=ptime:20
  326. a=sendrecv
  327. m=video 0 RTP/AVP 103 102 99
  328. ---
  329. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:3864 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 89.209.100.91:22980
  330. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:4108 retrans_pkt: SIP TIMER: Rescheduling retransmission #93531 (1) INVITE - 5
  331. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:4128 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 418 ms (t1 209 ms (Retrans id #93531))
  332. Retransmitting #1 (NAT) to 89.209.100.91:4481:
  333. INVITE sip:linphone.iphone@192.168.3.180:4481 SIP/2.0
  334. Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK014e74de;rport
  335. Max-Forwards: 70
  336. From: "Doctor" <sip:6000333@209.239.114.51>;tag=as4436c5b0
  337. To: <sip:6000444@89.209.100.91:4481;line=14bc6bcb7ab78b7>;tag=1613855871
  338. Contact: <sip:6000333@209.239.114.51:5060>
  339. Call-ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  340. CSeq: 105 INVITE
  341. User-Agent: Video Medicine PBX
  342. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  343. Supported: replaces, timer
  344. X-asterisk-Info: SIP re-invite (External RTP bridge)
  345. Content-Type: application/sdp
  346. Content-Length: 308
  347.  
  348. v=0
  349. o=root 1538395523 1538395526 IN IP4 192.168.3.96
  350. s=Asterisk PBX 11.2.1
  351. c=IN IP4 192.168.3.96
  352. t=0 0
  353. m=audio 7076 RTP/AVP 0 9 3 101
  354. a=rtpmap:0 PCMU/8000
  355. a=rtpmap:9 G722/8000
  356. a=rtpmap:3 GSM/8000
  357. a=rtpmap:101 telephone-event/8000
  358. a=fmtp:101 0-16
  359. a=silenceSupp:off - - - -
  360. a=ptime:20
  361. a=sendrecv
  362.  
  363. ---
  364. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:3864 __sip_xmit: Trying to put 'INVITE sip:' onto UDP socket destined for 89.209.100.91:4481
  365.  
  366. <--- SIP read from UDP:89.209.100.91:4481 --->
  367. SIP/2.0 100 Trying
  368. Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK014e74de;rport=5060
  369. From: "Doctor" <sip:6000333@209.239.114.51>;tag=as4436c5b0
  370. To: <sip:6000444@89.209.100.91:4481;line=14bc6bcb7ab78b7>;tag=1613855871
  371. Call-ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  372. CSeq: 105 INVITE
  373. User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  374. Content-Length: 0
  375.  
  376. <------------->
  377. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 0 [ 18]: SIP/2.0 100 Trying
  378. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK014e74de;rport=5060
  379. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" <sip:6000333@209.239.114.51>;tag=as4436c5b0
  380. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 3 [ 72]: To: <sip:6000444@89.209.100.91:4481;line=14bc6bcb7ab78b7>;tag=1613855871
  381. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  382. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 105 INVITE
  383. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 6 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  384. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 7 [ 17]: Content-Length: 0
  385. --- (8 headers 0 lines) ---
  386. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9118 find_call: = Looking for Call ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060 (Checking To) --From tag as4436c5b0 --To-tag 1613855871
  387. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000035c] bound to thread.
  388. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:4590 __sip_semi_ack: *** SIP TIMER: Cancelling retransmission #93531 - INVITE (got response)
  389. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:4597 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060' Request 105: Found
  390. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:22342 handle_response_invite: SIP response 100 to RE-invite on outgoing call 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  391. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000035c] being removed from thread.
  392.  
  393. <--- SIP read from UDP:89.209.100.91:22980 --->
  394. ACK sip:6000444@209.239.114.51:5060 SIP/2.0
  395. Via: SIP/2.0/UDP 192.168.3.96:22980;rport;branch=z9hG4bK1372128609
  396. From: <sip:6000333@vm.intersog.com:22071>;tag=1351731604
  397. To: "Patient" <sip:6000444@209.239.114.51>;tag=as7db15ebe
  398. Call-ID: 291853999
  399. CSeq: 22 ACK
  400. Contact: <sip:linphone.iphone@89.209.100.91:22980>
  401. Max-Forwards: 70
  402. User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  403. Content-Length: 0
  404.  
  405. <------------->
  406. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 0 [ 43]: ACK sip:6000444@209.239.114.51:5060 SIP/2.0
  407. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.3.96:22980;rport;branch=z9hG4bK1372128609
  408. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: <sip:6000333@vm.intersog.com:22071>;tag=1351731604
  409. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 3 [ 57]: To: "Patient" <sip:6000444@209.239.114.51>;tag=as7db15ebe
  410. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 4 [ 18]: Call-ID: 291853999
  411. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 5 [ 12]: CSeq: 22 ACK
  412. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 6 [ 50]: Contact: <sip:linphone.iphone@89.209.100.91:22980>
  413. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70
  414. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  415. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 9 [ 17]: Content-Length: 0
  416. --- (10 headers 0 lines) ---
  417. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9118 find_call: = Looking for Call ID: 291853999 (Checking From) --From tag 1351731604 --To-tag as7db15ebe
  418. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000035c] bound to thread.
  419. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:27866 handle_incoming: **** Received ACK (6) - Command in SIP ACK
  420. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:4523 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #93530
  421. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '291853999' of Response 22: Match Found
  422. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000035c] being removed from thread.
  423.  
  424. <--- SIP read from UDP:89.209.100.91:4481 --->
  425. SIP/2.0 200 OK
  426. Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK014e74de;rport=5060
  427. From: "Doctor" <sip:6000333@209.239.114.51>;tag=as4436c5b0
  428. To: <sip:6000444@89.209.100.91:4481;line=14bc6bcb7ab78b7>;tag=1613855871
  429. Call-ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  430. CSeq: 105 INVITE
  431. Contact: <sip:linphone.iphone@192.168.3.180:4481>
  432. Content-Type: application/sdp
  433. User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  434. Content-Length: 244
  435.  
  436. v=0
  437. o=6000444 622 2567 IN IP4 192.168.3.180
  438. s=Talk
  439. c=IN IP4 192.168.3.180
  440. b=AS:380
  441. t=0 0
  442. m=audio 7076 RTP/AVP 0 9 3 101
  443. a=rtpmap:0 PCMU/8000
  444. a=rtpmap:9 G722/8000
  445. a=rtpmap:3 GSM/8000
  446. a=rtpmap:101 telephone-event/8000
  447. a=fmtp:101 0-11
  448. <------------->
  449. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 0 [ 14]: SIP/2.0 200 OK
  450. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 1 [ 70]: Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK014e74de;rport=5060
  451. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 2 [ 58]: From: "Doctor" <sip:6000333@209.239.114.51>;tag=as4436c5b0
  452. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 3 [ 72]: To: <sip:6000444@89.209.100.91:4481;line=14bc6bcb7ab78b7>;tag=1613855871
  453. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 4 [ 61]: Call-ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  454. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 5 [ 16]: CSeq: 105 INVITE
  455. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 6 [ 49]: Contact: <sip:linphone.iphone@192.168.3.180:4481>
  456. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 7 [ 29]: Content-Type: application/sdp
  457. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  458. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 9 [ 19]: Content-Length: 244
  459. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 10 [ 0]:
  460. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 0 [ 3]: v=0
  461. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 1 [ 39]: o=6000444 622 2567 IN IP4 192.168.3.180
  462. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 2 [ 6]: s=Talk
  463. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 3 [ 22]: c=IN IP4 192.168.3.180
  464. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 4 [ 8]: b=AS:380
  465. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 5 [ 5]: t=0 0
  466. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 6 [ 30]: m=audio 7076 RTP/AVP 0 9 3 101
  467. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 7 [ 20]: a=rtpmap:0 PCMU/8000
  468. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 8 [ 20]: a=rtpmap:9 G722/8000
  469. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 9 [ 19]: a=rtpmap:3 GSM/8000
  470. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Body 10 [ 33]: a=rtpmap:101 telephone-event/8000
  471. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9605 parse_request: Body 11 [ 15]: a=fmtp:101 0-11
  472. --- (10 headers 12 lines) ---
  473. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9118 find_call: = Looking for Call ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060 (Checking To) --From tag as4436c5b0 --To-tag 1613855871
  474. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000035c] bound to thread.
  475. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:4518 __sip_ack: Acked pending invite 105
  476. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:4556 __sip_ack: Stopping retransmission on '0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060' of Request 105: Match Found
  477. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:22342 handle_response_invite: SIP response 200 to RE-invite on outgoing call 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  478. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
  479. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP o=6000444 622 2567 IN IP4 192.168.3.180... OK.
  480. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED.
  481. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.180' into...
  482. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.180' and port ''.
  483. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP c=IN IP4 192.168.3.180... OK.
  484. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP b=AS:380... UNSUPPORTED OR FAILED.
  485. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10000 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
  486. Found RTP audio format 0
  487. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 0 based on m type on 0x7fa82006fad0
  488. Found RTP audio format 9
  489. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 9 based on m type on 0x7fa82006fad0
  490. Found RTP audio format 3
  491. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 3 based on m type on 0x7fa82006fad0
  492. Found RTP audio format 101
  493. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:557 ast_rtp_codecs_payloads_set_m_type: Setting payload 101 based on m type on 0x7fa82006fad0
  494. Found audio description format PCMU for ID 0
  495. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
  496. Found audio description format G722 for ID 9
  497. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
  498. Found audio description format GSM for ID 3
  499. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
  500. Found audio description format telephone-event for ID 101
  501. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
  502. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10406 process_sdp: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED OR FAILED.
  503. Capabilities: us - (gsm|ulaw|g729|g722|h263|h263p|h264|mpeg4|vp8), peer - audio=(gsm|ulaw|g722)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|g722)
  504. Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
  505. Peer audio RTP is at port 192.168.3.180:7076
  506. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 0 from 0x7fa82006fad0 to 0x7fa8240889b0
  507. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 3 from 0x7fa82006fad0 to 0x7fa8240889b0
  508. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 9 from 0x7fa82006fad0 to 0x7fa8240889b0
  509. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: rtp_engine.c:515 ast_rtp_codecs_payloads_copy: Copying payload 101 from 0x7fa82006fad0 to 0x7fa8240889b0
  510. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: res_rtp_asterisk.c:3851 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7fa8240887e8'
  511. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into...
  512. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''.
  513. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '209.239.114.51' into...
  514. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '209.239.114.51' and port ''.
  515. Peer doesn't provide video
  516. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: res_rtp_asterisk.c:3896 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7fa8240be968'
  517. Peer doesn't provide T.140
  518. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10668 process_sdp: We're settling with these formats: (gsm|ulaw|g722)
  519. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10675 process_sdp: We have an owner, now see if we need to change this call
  520. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:10681 process_sdp: Setting native formats after processing SDP. peer joint formats (gsm|ulaw|g722), old nativeformats (ulaw)
  521. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:6669 update_call_counter: Updating call counter for outgoing call
  522. [Mar 29 10:34:34] DEBUG[1030]: devicestate.c:344 _ast_device_state: No provider found, checking channel drivers for SIP - 6000444
  523. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:11802 reqprep: Strict routing enforced for session 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  524. [Mar 29 10:34:34] DEBUG[1030]: chan_sip.c:29297 sip_devicestate: Checking device state for peer 6000444
  525. set_destination: Parsing <sip:linphone.iphone@192.168.3.180:4481> for address/port to send to
  526. [Mar 29 10:34:34] DEBUG[1030]: devicestate.c:467 do_state_change: Changing state for SIP/6000444 - state 2 (In use)
  527. [Mar 29 10:34:34] DEBUG[1030]: devicestate.c:442 devstate_event: device 'SIP/6000444' state '2'
  528. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '192.168.3.180:4481' into...
  529. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: netsock2.c:192 ast_sockaddr_split_hostport: ...host '192.168.3.180' and port '4481'.
  530. set_destination: set destination to 192.168.3.180:4481
  531. [Mar 29 10:34:34] DEBUG[1266]: app_queue.c:1804 handle_statechange: Device 'SIP/6000444' changed to state '2' (In use) but we don't care because they're not a member of any queue.
  532. Transmitting (NAT) to 89.209.100.91:4481:
  533. ACK sip:linphone.iphone@192.168.3.180:4481 SIP/2.0
  534. Via: SIP/2.0/UDP 209.239.114.51:5060;branch=z9hG4bK4d2baab1;rport
  535. Max-Forwards: 70
  536. From: "Doctor" <sip:6000333@209.239.114.51>;tag=as4436c5b0
  537. To: <sip:6000444@89.209.100.91:4481;line=14bc6bcb7ab78b7>;tag=1613855871
  538. Contact: <sip:6000333@209.239.114.51:5060>
  539. Call-ID: 0d81346b38d76b086fdee2be17a7393f@209.239.114.51:5060
  540. CSeq: 105 ACK
  541. User-Agent: Video Medicine PBX
  542. Content-Length: 0
  543.  
  544.  
  545. ---
  546. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:3864 __sip_xmit: Trying to put 'ACK sip:lin' onto UDP socket destined for 89.209.100.91:4481
  547. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000035c] being removed from thread.
  548.  
  549. <--- SIP read from UDP:89.209.100.91:22980 --->
  550. ACK sip:6000444@209.239.114.51:5060 SIP/2.0
  551. Via: SIP/2.0/UDP 192.168.3.96:22980;rport;branch=z9hG4bK1372128609
  552. From: <sip:6000333@vm.intersog.com:22071>;tag=1351731604
  553. To: "Patient" <sip:6000444@209.239.114.51>;tag=as7db15ebe
  554. Call-ID: 291853999
  555. CSeq: 22 ACK
  556. Contact: <sip:linphone.iphone@89.209.100.91:22980>
  557. Max-Forwards: 70
  558. User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  559. Content-Length: 0
  560.  
  561. <------------->
  562. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 0 [ 43]: ACK sip:6000444@209.239.114.51:5060 SIP/2.0
  563. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 1 [ 66]: Via: SIP/2.0/UDP 192.168.3.96:22980;rport;branch=z9hG4bK1372128609
  564. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 2 [ 56]: From: <sip:6000333@vm.intersog.com:22071>;tag=1351731604
  565. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 3 [ 57]: To: "Patient" <sip:6000444@209.239.114.51>;tag=as7db15ebe
  566. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 4 [ 18]: Call-ID: 291853999
  567. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 5 [ 12]: CSeq: 22 ACK
  568. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 6 [ 50]: Contact: <sip:linphone.iphone@89.209.100.91:22980>
  569. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 7 [ 16]: Max-Forwards: 70
  570. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 8 [ 48]: User-Agent: LinphoneIPhone/2.0.2 (eXosip2/3.6.0)
  571. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9568 parse_request: Header 9 [ 17]: Content-Length: 0
  572. --- (10 headers 0 lines) ---
  573. [Mar 29 10:34:34] DEBUG[1238]: chan_sip.c:9118 find_call: = Looking for Call ID: 291853999 (Checking From) --From tag 1351731604 --To-tag as7db15ebe
  574. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1324 ast_callid_threadassoc_add: CALL_ID [C-0000035c] bound to thread.
  575. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: chan_sip.c:27866 handle_incoming: **** Received ACK (6) - Command in SIP ACK
  576. [Mar 29 10:34:34] DEBUG[1238][C-0000035c]: logger.c:1346 ast_callid_threadassoc_remove: Call_ID [C-0000035c] being removed from thread.
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