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SIP DEBUG

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Jun 28th, 2011
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  1. INVITE sip:[email protected] SIP/2.0
  2. Record-Route: <sip:1.2.3.4;lr;ftag=ea220aee7e819bfdo1;nat=yes>
  3. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK67ca.231c1751.0
  4. Via: SIP/2.0/UDP 5.6.7.8:25267;rport=1065;received=5.6.7.8;branch=z9hG4bK-eda45f00
  5. From: "CALLER" <sip:[email protected]>;tag=ea220aee7e819bfdo1
  6. To: <sip:[email protected]>
  7. Call-ID: [email protected]
  8. CSeq: 102 INVITE
  9. Max-Forwards: 69
  10. Contact: "CALLER" <sip:[email protected]:1065>
  11. Expires: 240
  12. User-Agent: Linksys/SPA962-6.1.5(a)
  13. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
  14. Supported: replaces
  15. Content-Type: application/sdp
  16. Content-Length: 229
  17. P-Asserted-Identity: <sip:[email protected]>
  18. Privacy: none
  19.  
  20. v=0
  21. o=- 20948884 20948884 IN IP4 5.6.7.8
  22. s=-
  23. c=IN IP4 1.2.3.4
  24. t=0 0
  25. m=audio 55932 RTP/AVP 8 101
  26. a=rtpmap:8 PCMA/8000
  27. a=rtpmap:101 telephone-event/8000
  28. a=fmtp:101 0-15
  29. a=ptime:30
  30. a=sendrecv
  31. a=nortpproxy:yes
  32.  
  33.  
  34.  
  35. <------------->
  36. [Jun 27 18:45:32] VERBOSE[2902] logger.c: --- (18 headers 12 lines) ---
  37. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Sending to 1.2.3.4 : 5060 (no NAT)
  38. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Using INVITE request as basis request - [email protected]
  39. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Found peer 'proxy-1'
  40. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Found RTP audio format 8
  41. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Found RTP audio format 101
  42. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Found audio description format PCMA for ID 8
  43. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Found audio description format telephone-event for ID 101
  44. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
  45. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  46. [Jun 27 18:45:32] DEBUG[2902] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
  47. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Peer audio RTP is at port 1.2.3.4:55932
  48. [Jun 27 18:45:32] VERBOSE[2902] logger.c: Looking for CALLEE in from-proxy (domain my.sipproxy.it)
  49. [Jun 27 18:45:32] VERBOSE[2902] logger.c: list_route: hop: <sip:1.2.3.4;lr;ftag=ea220aee7e819bfdo1;nat=yes>
  50. [Jun 27 18:45:32] VERBOSE[2902] logger.c:
  51. <--- Transmitting (no NAT) to 1.2.3.4:5060 --->
  52. SIP/2.0 100 Trying
  53. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK67ca.231c1751.0;received=1.2.3.4
  54. Via: SIP/2.0/UDP 5.6.7.8:25267;rport=1065;received=5.6.7.8;branch=z9hG4bK-eda45f00
  55. Record-Route: <sip:1.2.3.4;lr;ftag=ea220aee7e819bfdo1;nat=yes>
  56. From: "CALLER" <sip:[email protected]>;tag=ea220aee7e819bfdo1
  57. To: <sip:[email protected]>
  58. Call-ID: [email protected]
  59. CSeq: 102 INVITE
  60. User-Agent: OVSoftSwitch-2.0
  61. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  62. Supported: replaces
  63. Contact: <sip:[email protected]>
  64. Content-Length: 0
  65.  
  66.  
  67.  
  68. INVITE sip:[email protected] SIP/2.0
  69. Via: SIP/2.0/UDP 9.10.11.12:5060;branch=z9hG4bK41060677;rport
  70. From: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  71. To: <sip:[email protected]>
  72. Contact: <sip:[email protected]>
  73. Call-ID: [email protected]
  74. CSeq: 102 INVITE
  75. User-Agent: OVSoftSwitch-2.0
  76. Max-Forwards: 70
  77. Date: Mon, 27 Jun 2011 16:45:32 GMT
  78. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  79. Supported: replaces
  80. P-Asserted-Identity: <sip:[email protected]>
  81. Content-Type: application/sdp
  82. Content-Length: 279
  83.  
  84. v=0
  85. o=root 2877 2877 IN IP4 9.10.11.12
  86. s=session
  87. c=IN IP4 9.10.11.12
  88. t=0 0
  89. m=audio 16682 RTP/AVP 8 18 101
  90. a=rtpmap:8 PCMA/8000
  91. a=rtpmap:18 G729/8000
  92. a=fmtp:18 annexb=no
  93. a=rtpmap:101 telephone-event/8000
  94. a=fmtp:101 0-16
  95. a=silenceSupp:off - - - -
  96. a=ptime:20
  97. a=sendrecv
  98.  
  99.  
  100.  
  101. ---
  102. [Jun 27 18:45:32] VERBOSE[7836] logger.c: -- Called BT-OUT-Trunk-1/CALLEE
  103. [Jun 27 18:45:32] VERBOSE[2902] logger.c:
  104. <--- SIP read from 13.14.15.16:5060 --->
  105. SIP/2.0 100 Trying
  106. Via: SIP/2.0/UDP 9.10.11.12:5060;branch=z9hG4bK41060677;rport=5060
  107. From: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  108. To: <sip:[email protected]>
  109. Call-ID: [email protected]
  110. CSeq: 102 INVITE
  111.  
  112.  
  113. <------------->
  114. [Jun 27 18:45:32] VERBOSE[2902] logger.c: --- (6 headers 0 lines) ---
  115. [Jun 27 18:45:33] VERBOSE[2902] logger.c:
  116. <--- SIP read from 13.14.15.16:5060 --->
  117. SIP/2.0 183 Session Progress
  118. Via: SIP/2.0/UDP 9.10.11.12:5060;branch=z9hG4bK41060677;rport=5060
  119. From: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  120. To: <sip:[email protected]>;tag=SDmnt8199-1c493502
  121. Call-ID: [email protected]
  122. CSeq: 102 INVITE
  123. Supported: replaces
  124. Contact: <sip:[email protected]:5060;transport=udp>
  125. Content-Type: application/sdp
  126. Content-Length: 154
  127.  
  128. v=0
  129. o=- 0 0 IN IP4 13.14.15.16
  130. s=IMSS
  131. c=IN IP4 13.14.15.16
  132. t=0 0
  133. m=audio 14988 RTP/AVP 18 101
  134. a=rtpmap:101 telephone-event/8000
  135. a=fmtp:101 0-15
  136.  
  137.  
  138.  
  139.  
  140. <------------->
  141. [Jun 27 18:45:33] VERBOSE[2902] logger.c: --- (10 headers 8 lines) ---
  142. [Jun 27 18:45:33] VERBOSE[2902] logger.c: Found RTP audio format 18
  143. [Jun 27 18:45:33] VERBOSE[2902] logger.c: Found RTP audio format 101
  144. [Jun 27 18:45:33] VERBOSE[2902] logger.c: Found audio description format telephone-event for ID 101
  145. [Jun 27 18:45:33] VERBOSE[2902] logger.c: Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
  146. [Jun 27 18:45:33] VERBOSE[2902] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  147. [Jun 27 18:45:33] DEBUG[2902] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
  148. [Jun 27 18:45:33] VERBOSE[2902] logger.c: Peer audio RTP is at port 13.14.15.16:14988
  149. [Jun 27 18:45:33] VERBOSE[7836] logger.c: -- SIP/BT-OUT-Trunk-1-0009385d is making progress passing it to SIP/proxy-1-0009385c
  150. [Jun 27 18:45:33] VERBOSE[7836] logger.c: Audio is at 17.18.19.20 port 17746
  151. [Jun 27 18:45:33] VERBOSE[7836] logger.c: Adding codec 0x8 (alaw) to SDP
  152. [Jun 27 18:45:33] VERBOSE[7836] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
  153. [Jun 27 18:45:33] VERBOSE[7836] logger.c:
  154. <--- Transmitting (no NAT) to 1.2.3.4:5060 --->
  155. SIP/2.0 183 Session Progress
  156. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK67ca.231c1751.0;received=1.2.3.4
  157. Via: SIP/2.0/UDP 5.6.7.8:25267;rport=1065;received=5.6.7.8;branch=z9hG4bK-eda45f00
  158. Record-Route: <sip:1.2.3.4;lr;ftag=ea220aee7e819bfdo1;nat=yes>
  159. From: "CALLER" <sip:[email protected]>;tag=ea220aee7e819bfdo1
  160. To: <sip:[email protected]>;tag=as697fdd56
  161. Call-ID: [email protected]
  162. CSeq: 102 INVITE
  163. User-Agent: OVSoftSwitch-2.0
  164. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  165. Supported: replaces
  166. Contact: <sip:[email protected]>
  167. Content-Type: application/sdp
  168. Content-Length: 242
  169.  
  170. v=0
  171. o=root 2877 2877 IN IP4 17.18.19.20
  172. s=session
  173. c=IN IP4 17.18.19.20
  174. t=0 0
  175. m=audio 17746 RTP/AVP 8 101
  176. a=rtpmap:8 PCMA/8000
  177. a=rtpmap:101 telephone-event/8000
  178. a=fmtp:101 0-16
  179. a=silenceSupp:off - - - -
  180. a=ptime:20
  181. a=sendrecv
  182.  
  183.  
  184. [Jun 27 18:45:34] VERBOSE[2902] logger.c:
  185. <--- SIP read from 13.14.15.16:5060 --->
  186. SIP/2.0 180 Ringing
  187. Via: SIP/2.0/UDP 9.10.11.12:5060;branch=z9hG4bK41060677;rport=5060
  188. From: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  189. To: <sip:[email protected]>;tag=SDmnt8199-1c493502
  190. Call-ID: [email protected]
  191. CSeq: 102 INVITE
  192. Supported: replaces
  193. Contact: <sip:[email protected]:5060;transport=udp>
  194. Content-Length: 0
  195.  
  196.  
  197. <------------->
  198. [Jun 27 18:45:34] VERBOSE[2902] logger.c: --- (9 headers 0 lines) ---
  199. [Jun 27 18:45:34] VERBOSE[7836] logger.c: -- SIP/BT-OUT-Trunk-1-0009385d is ringing
  200. [Jun 27 18:45:34] VERBOSE[7836] logger.c:
  201. <--- Transmitting (no NAT) to 1.2.3.4:5060 --->
  202. SIP/2.0 180 Ringing
  203. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK67ca.231c1751.0;received=1.2.3.4
  204. Via: SIP/2.0/UDP 5.6.7.8:25267;rport=1065;received=5.6.7.8;branch=z9hG4bK-eda45f00
  205. Record-Route: <sip:1.2.3.4;lr;ftag=ea220aee7e819bfdo1;nat=yes>
  206. From: "CALLER" <sip:[email protected]>;tag=ea220aee7e819bfdo1
  207. To: <sip:[email protected]>;tag=as697fdd56
  208. Call-ID: [email protected]
  209. CSeq: 102 INVITE
  210. User-Agent: OVSoftSwitch-2.0
  211. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  212. Supported: replaces
  213. Contact: <sip:[email protected]>
  214. Content-Length: 0
  215.  
  216.  
  217. <------------>
  218. [Jun 27 18:45:34] VERBOSE[2902] logger.c:
  219. <--- SIP read from 13.14.15.16:5060 --->
  220. SIP/2.0 200 OK
  221. Via: SIP/2.0/UDP 9.10.11.12:5060;branch=z9hG4bK41060677;rport=5060
  222. From: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  223. To: <sip:[email protected]>;tag=SDmnt8199-1c493502
  224. Call-ID: [email protected]
  225. CSeq: 102 INVITE
  226. Contact: <sip:[email protected]:5060;transport=udp>
  227. Content-Type: application/sdp
  228. Content-Length: 154
  229.  
  230. v=0
  231. o=- 0 0 IN IP4 13.14.15.16
  232. s=IMSS
  233. c=IN IP4 13.14.15.16
  234. t=0 0
  235. m=audio 14988 RTP/AVP 18 101
  236. a=rtpmap:101 telephone-event/8000
  237. a=fmtp:101 0-15
  238.  
  239. <------------->
  240. [Jun 27 18:45:34] VERBOSE[2902] logger.c: --- (9 headers 8 lines) ---
  241. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Found RTP audio format 18
  242. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Found RTP audio format 101
  243. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Found audio description format telephone-event for ID 101
  244. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Capabilities: us - 0x108 (alaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
  245. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  246. [Jun 27 18:45:34] DEBUG[2902] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
  247. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Peer audio RTP is at port 13.14.15.16:14988
  248. [Jun 27 18:45:34] VERBOSE[2902] logger.c: list_route: hop: <sip:[email protected]:5060;transport=udp>
  249. [Jun 27 18:45:34] DEBUG[2902] chan_sip.c: Strict routing enforced for session [email protected]
  250. [Jun 27 18:45:34] VERBOSE[2902] logger.c: set_destination: Parsing <sip:[email protected]:5060;transport=udp> for address/port to send to
  251. [Jun 27 18:45:34] VERBOSE[2902] logger.c: set_destination: set destination to 13.14.15.16, port 5060
  252. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Transmitting (no NAT) to 13.14.15.16:5060:
  253.  
  254. ACK sip:[email protected]:5060;transport=udp SIP/2.0
  255. Via: SIP/2.0/UDP 9.10.11.12:5060;branch=z9hG4bK16168add;rport
  256. From: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  257. To: <sip:[email protected]>;tag=SDmnt8199-1c493502
  258. Contact: <sip:[email protected]>
  259. Call-ID: [email protected]
  260. CSeq: 102 ACK
  261. User-Agent: OVSoftSwitch-2.0
  262. Max-Forwards: 70
  263. Content-Length: 0
  264.  
  265.  
  266. ---
  267. [Jun 27 18:45:34] VERBOSE[7836] logger.c: -- SIP/BT-OUT-Trunk-1-0009385d answered SIP/proxy-1-0009385c
  268. [Jun 27 18:45:34] VERBOSE[7836] logger.c: Audio is at 17.18.19.20 port 17746
  269. [Jun 27 18:45:34] VERBOSE[7836] logger.c: Adding codec 0x8 (alaw) to SDP
  270. [Jun 27 18:45:34] VERBOSE[7836] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
  271. [Jun 27 18:45:34] VERBOSE[7836] logger.c:
  272. <--- Reliably Transmitting (no NAT) to 1.2.3.4:5060 --->
  273. SIP/2.0 200 OK
  274. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK67ca.231c1751.0;received=1.2.3.4
  275. Via: SIP/2.0/UDP 5.6.7.8:25267;rport=1065;received=5.6.7.8;branch=z9hG4bK-eda45f00
  276. Record-Route: <sip:1.2.3.4;lr;ftag=ea220aee7e819bfdo1;nat=yes>
  277. From: "CALLER" <sip:[email protected]>;tag=ea220aee7e819bfdo1
  278. To: <sip:[email protected]>;tag=as697fdd56
  279. Call-ID: [email protected]
  280. CSeq: 102 INVITE
  281. User-Agent: OVSoftSwitch-2.0
  282. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  283. Supported: replaces
  284. Contact: <sip:[email protected]>
  285. Content-Type: application/sdp
  286. Content-Length: 242
  287.  
  288. v=0
  289. o=root 2877 2878 IN IP4 17.18.19.20
  290. s=session
  291. c=IN IP4 17.18.19.20
  292. t=0 0
  293. m=audio 17746 RTP/AVP 8 101
  294. a=rtpmap:8 PCMA/8000
  295. a=rtpmap:101 telephone-event/8000
  296. a=fmtp:101 0-16
  297. a=silenceSupp:off - - - -
  298. a=ptime:20
  299. a=sendrecv
  300.  
  301.  
  302. <------------>
  303. [Jun 27 18:45:34] VERBOSE[2902] logger.c:
  304. <--- SIP read from 1.2.3.4:5060 --->
  305. ACK sip:[email protected] SIP/2.0
  306. Record-Route: <sip:1.2.3.4;lr;ftag=ea220aee7e819bfdo1;nat=yes>
  307. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK67ca.231c1751.2
  308. Via: SIP/2.0/UDP 5.6.7.8:25267;rport=1065;received=5.6.7.8;branch=z9hG4bK-10d407fa
  309. From: "CALLER" <sip:[email protected]>;tag=ea220aee7e819bfdo1
  310. To: <sip:[email protected]>;tag=as697fdd56
  311. Call-ID: [email protected]
  312. CSeq: 102 ACK
  313. Max-Forwards: 69
  314. Proxy-Authorization: Digest username="CALLER",realm="my.sipproxy.it",nonce="4e08b4d80000e55946dc20e533f75e5539df72608eb72711",uri="sip:[email protected]",algorithm=MD5,response="fdf4fecfa39254e10f88b0a8c52bdb78"
  315. Contact: "CALLER" <sip:[email protected]:1065>
  316. User-Agent: Linksys/SPA962-6.1.5(a)
  317. Content-Length: 0
  318. Privacy: none
  319.  
  320.  
  321. <------------->
  322. [Jun 27 18:45:34] VERBOSE[2902] logger.c: --- (14 headers 0 lines) ---
  323. [Jun 27 18:45:34] VERBOSE[2902] logger.c:
  324. <--- SIP read from 13.14.15.16:5060 --->
  325. INVITE sip:[email protected] SIP/2.0
  326. Via: SIP/2.0/UDP 13.14.15.16:5060;branch=z9hG4bKds95s7109oph2ksoj6s0sb0000gj1.1
  327. To: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  328. From: <sip:[email protected]>;tag=SDmnt8199-1c493502
  329. Call-ID: [email protected]
  330. CSeq: 103 INVITE
  331. Max-Forwards: 68
  332. Contact: <sip:[email protected]:5060;transport=udp>
  333. Supported: replaces
  334. P-Asserted-Identity: <sip:[email protected]>
  335. Content-Type: application/sdp
  336. Content-Length: 219
  337.  
  338. v=0
  339. o=- 0 0 IN IP4 13.14.15.16
  340. s=IMSS
  341. c=IN IP4 13.14.15.16
  342. t=0 0
  343. m=audio 14988 RTP/AVP 8 101 103
  344. a=rtpmap:101 telephone-event/8000
  345. a=rtpmap:103 X-NSE/8000
  346. a=fmtp:101 0-15
  347. a=fmtp:103 0-15,192-198,200-202,204
  348.  
  349. <------------->
  350. [Jun 27 18:45:34] VERBOSE[2902] logger.c: --- (12 headers 10 lines) ---
  351. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Sending to 13.14.15.16 : 5060 (no NAT)
  352. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Found RTP audio format 8
  353. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Found RTP audio format 101
  354. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Found RTP audio format 103
  355. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Found audio description format telephone-event for ID 101
  356. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Found unknown media description format X-NSE for ID 103
  357. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Capabilities: us - 0x108 (alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
  358. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
  359. [Jun 27 18:45:34] DEBUG[2902] chan_sip.c: Our T38 capability = (0), peer T38 capability (0), joint T38 capability (0)
  360. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Peer audio RTP is at port 13.14.15.16:14988
  361. [Jun 27 18:45:34] DEBUG[2902] chan_sip.c: Oooh, we need to change our audio formats since our peer supports only 0x8 (alaw) and not 0x100 (g729)
  362. [Jun 27 18:45:34] VERBOSE[2902] logger.c:
  363. <--- Transmitting (no NAT) to 13.14.15.16:5060 --->
  364. SIP/2.0 100 Trying
  365. Via: SIP/2.0/UDP 13.14.15.16:5060;branch=z9hG4bKds95s7109oph2ksoj6s0sb0000gj1.1;received=13.14.15.16
  366. From: <sip:[email protected]>;tag=SDmnt8199-1c493502
  367. To: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  368. Call-ID: [email protected]
  369. CSeq: 103 INVITE
  370. User-Agent: OVSoftSwitch-2.0
  371. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  372. Supported: replaces
  373. Contact: <sip:[email protected]>
  374. Content-Length: 0
  375.  
  376.  
  377. <------------>
  378. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Audio is at 9.10.11.12 port 16682
  379. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Adding codec 0x8 (alaw) to SDP
  380. [Jun 27 18:45:34] VERBOSE[2902] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
  381. [Jun 27 18:45:34] VERBOSE[2902] logger.c:
  382. <--- Reliably Transmitting (no NAT) to 13.14.15.16:5060 --->
  383. SIP/2.0 200 OK
  384. Via: SIP/2.0/UDP 13.14.15.16:5060;branch=z9hG4bKds95s7109oph2ksoj6s0sb0000gj1.1;received=13.14.15.16
  385. From: <sip:[email protected]>;tag=SDmnt8199-1c493502
  386. To: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  387. Call-ID: [email protected]
  388. CSeq: 103 INVITE
  389. User-Agent: OVSoftSwitch-2.0
  390. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  391. Supported: replaces
  392. Contact: <sip:[email protected]>
  393. Content-Type: application/sdp
  394. Content-Length: 232
  395.  
  396. v=0
  397. o=root 2877 2878 IN IP4 9.10.11.12
  398. s=session
  399. c=IN IP4 9.10.11.12
  400. t=0 0
  401. m=audio 16682 RTP/AVP 8 101
  402. a=rtpmap:8 PCMA/8000
  403. a=rtpmap:101 telephone-event/8000
  404. a=fmtp:101 0-16
  405. a=silenceSupp:off - - - -
  406. a=ptime:20
  407. a=sendrecv
  408.  
  409. <------------>
  410. [Jun 27 18:45:34] VERBOSE[2902] logger.c:
  411. <--- SIP read from 13.14.15.16:5060 --->
  412. ACK sip:[email protected] SIP/2.0
  413. Via: SIP/2.0/UDP 13.14.15.16:5060;branch=z9hG4bKe4n90q1018604ksl96k0.1
  414. To: "CALLER" <sip:[email protected]>;tag=as38ac8b3a
  415. From: <sip:[email protected]>;tag=SDmnt8199-1c493502
  416. Call-ID: [email protected]
  417. CSeq: 103 ACK
  418. Max-Forwards: 68
  419. Content-Length: 0
  420.  
  421.  
  422. At this time the UAC does not hear audio at all.
  423.  
  424.  
  425. <--- SIP read from 1.2.3.4:5060 --->
  426. BYE sip:[email protected] SIP/2.0
  427. Record-Route: <sip:1.2.3.4;lr;ftag=ea220aee7e819bfdo1;nat=yes>
  428. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK77ca.29fa4073.0
  429. Via: SIP/2.0/UDP 5.6.7.8:25267;rport=1065;received=5.6.7.8;branch=z9hG4bK-7d0eb43e
  430. From: "CALLER" <sip:[email protected]>;tag=ea220aee7e819bfdo1
  431. To: <sip:[email protected]>;tag=as697fdd56
  432. Call-ID: [email protected]
  433. CSeq: 103 BYE
  434. Max-Forwards: 69
  435. Proxy-Authorization: Digest username="CALLER",realm="my.sipproxy.it",nonce="4e08b4d80000e55946dc20e533f75e5539df72608eb72711",uri="sip:[email protected]",algorithm=MD5,response="5bbd17d4bd2551683f6894d22e4a98f5"
  436. User-Agent: Linksys/SPA962-6.1.5(a)
  437. Content-Length: 0
  438. Privacy: none
  439.  
  440.  
  441. <------------->
  442. [Jun 27 18:45:42] VERBOSE[2902] logger.c: --- (13 headers 0 lines) ---
  443. [Jun 27 18:45:42] VERBOSE[2902] logger.c: Sending to 1.2.3.4 : 5060 (no NAT)
  444. [Jun 27 18:45:42] VERBOSE[2902] logger.c:
  445. <--- Transmitting (no NAT) to 1.2.3.4:5060 --->
  446. SIP/2.0 200 OK
  447. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK77ca.29fa4073.0;received=1.2.3.4
  448. Via: SIP/2.0/UDP 5.6.7.8:25267;rport=1065;received=5.6.7.8;branch=z9hG4bK-7d0eb43e
  449. Record-Route: <sip:1.2.3.4;lr;ftag=ea220aee7e819bfdo1;nat=yes>
  450. From: "CALLER" <sip:[email protected]>;tag=ea220aee7e819bfdo1
  451. To: <sip:[email protected]>;tag=as697fdd56
  452. Call-ID: [email protected]
  453. CSeq: 103 BYE
  454. User-Agent: OVSoftSwitch-2.0
  455. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
  456. Supported: replaces
  457. Content-Length: 0
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