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Asterisk + WebRTC subscribing UA jsSIP

powerponch
Sep 8th, 2015
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  1. REGISTER sip:192.168.0.4 SIP/2.0
  2. Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK9352120
  3. Max-Forwards: 69
  4. To: <sip:[email protected]>
  5. From: "UA WebRTC" <sip:[email protected]>;tag=f8m06houfb
  6. Call-ID: liamoji649fhds9jtdajfi
  7. CSeq: 1 REGISTER
  8. Contact: <sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:b1a09033-7533-47a8-a039-1a6760b2af84>";expires=600
  9. Expires: 600
  10. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
  11. Supported: path,gruu,outbound
  12. User-Agent: JsSIP 0.7.4
  13. Content-Length: 0
  14.  
  15. +15ms
  16. jssip-0.7.4.js (line 22459)
  17. JsSIP:Transport
  18. received WebSocket text message:
  19.  
  20. SIP/2.0 401 Unauthorized
  21. Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK9352120;received=192.168.0.4
  22. From: "UA WebRTC" <sip:[email protected]>;tag=f8m06houfb
  23. To: <sip:[email protected]>;tag=as2235a73d
  24. Call-ID: liamoji649fhds9jtdajfi
  25. CSeq: 1 REGISTER
  26. Server: Asterisk PBX 12.8.2
  27. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  28. Supported: replaces, timer
  29. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0f6c6c26"
  30. Content-Length: 0
  31.  
  32.  
  33. +30ms
  34. jssip-0.7.4.js (line 22459)
  35. JsSIP:Transport
  36. sending WebSocket message:
  37.  
  38. REGISTER sip:192.168.0.4 SIP/2.0
  39. Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK1900716
  40. Max-Forwards: 69
  41. To: <sip:[email protected]>
  42. From: "UA WebRTC" <sip:[email protected]>;tag=f8m06houfb
  43. Call-ID: liamoji649fhds9jtdajfi
  44. CSeq: 2 REGISTER
  45. Authorization: Digest algorithm=MD5, username="1001", realm="asterisk", nonce="0f6c6c26", uri="sip:192.168.0.4", response="bed4984f5fc3c21f3854dccdb30a74b2"
  46. Contact: <sip:[email protected];transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:b1a09033-7533-47a8-a039-1a6760b2af84>";expires=600
  47. Expires: 600
  48. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER
  49. Supported: path,gruu,outbound
  50. User-Agent: JsSIP 0.7.4
  51. Content-Length: 0
  52.  
  53. +32ms
  54. jssip-0.7.4.js (line 22459)
  55. JsSIP:Transport
  56. received WebSocket text message:
  57.  
  58. SIP/2.0 200 OK
  59. Via: SIP/2.0/WS 192.0.2.177;branch=z9hG4bK1900716;received=192.168.0.4
  60. From: "UA WebRTC" <sip:[email protected]>;tag=f8m06houfb
  61. To: <sip:[email protected]>;tag=as2235a73d
  62. Call-ID: liamoji649fhds9jtdajfi
  63. CSeq: 2 REGISTER
  64. Server: Asterisk PBX 12.8.2
  65. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  66.  
  67. Supported: replaces, timer
  68. Expires: 600
  69. Contact: <sip:[email protected];transport=ws>;expires=600
  70. Date: Mon, 24 Aug 2015 14:15:52 GMT
  71. Content-Length: 0
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