FreeCalypso > hg > themwi-system-sw
annotate sip-in/retrans.c @ 170:a6eb2de277f6
mgw: massive simplification for continuous RTP stream from BTS
author | Mychaela Falconia <falcon@freecalypso.org> |
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date | Sun, 20 Nov 2022 01:58:47 -0800 |
parents | bf3b19bf57e9 |
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rev | line source |
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1 /* |
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2 * In this module we handle retransmission of INVITE responses |
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3 * and BYE requests. |
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4 */ |
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5 |
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6 #include <sys/types.h> |
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7 #include <sys/socket.h> |
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8 #include <sys/time.h> |
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9 #include <netinet/in.h> |
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10 #include <stdio.h> |
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11 #include <stdint.h> |
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12 #include <stdlib.h> |
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13 #include <string.h> |
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14 #include <strings.h> |
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15 #include <syslog.h> |
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16 #include "../include/gsm48_const.h" |
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17 #include "../libsip/out_msg.h" |
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18 #include "call.h" |
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19 |
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20 extern unsigned cfg_retrans_count; |
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21 extern unsigned sip_linger_error; |
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22 extern struct call *call_list; |
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23 |
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24 void |
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25 run_periodic_retrans() |
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26 { |
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27 struct call *call; |
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28 struct sip_msg_out msg; |
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29 |
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30 for (call = call_list; call; call = call->next) { |
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31 switch (call->sip_state) { |
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32 case SIP_STATE_RINGING_REL: |
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33 if (call->sip_tx_count < cfg_retrans_count) { |
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34 start_response_out_msg(&msg, "180 Ringing"); |
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35 fill_invite_resp_from_call(&msg, call); |
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36 out_msg_add_header(&msg, "Require", "100rel"); |
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37 out_msg_add_header(&msg, "RSeq", "1"); |
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38 out_msg_finish(&msg); |
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39 sip_tx_packet(&msg, &call->udp_sin); |
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40 call->sip_tx_count++; |
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41 } else { |
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42 syslog(LOG_ERR, |
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43 "Call in%06u: 100rel retrans timeout", |
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44 call->in_tag_num); |
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45 call->overall_state = OVERALL_STATE_TEARDOWN; |
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46 disconnect_mncc(call, GSM48_CAUSE_LOC_PRN_S_LU, |
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47 GSM48_CC_CAUSE_INTERWORKING); |
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48 disconnect_tmgw(call); |
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49 strcpy(call->invite_fail, |
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50 "504 100rel retransmissions exhausted"); |
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51 signal_invite_error(call); |
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52 } |
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53 break; |
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54 case SIP_STATE_INVITE_200: |
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55 if (call->sip_tx_count < cfg_retrans_count) { |
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56 fill_invite_200_resp(&msg, call); |
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57 sip_tx_packet(&msg, &call->udp_sin); |
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58 call->sip_tx_count++; |
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59 } else { |
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60 syslog(LOG_ERR, |
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61 "Call in%06u: 200 retrans timeout", |
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62 call->in_tag_num); |
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63 call->overall_state = OVERALL_STATE_TEARDOWN; |
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64 disconnect_mncc(call, GSM48_CAUSE_LOC_PRN_S_LU, |
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65 GSM48_CC_CAUSE_INTERWORKING); |
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66 disconnect_tmgw(call); |
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67 initiate_bye(call); |
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68 } |
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69 break; |
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70 case SIP_STATE_INVITE_ERR: |
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71 if (call->sip_tx_count < cfg_retrans_count) { |
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72 start_response_out_msg(&msg, call->invite_fail); |
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73 fill_invite_resp_from_call(&msg, call); |
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74 out_msg_finish(&msg); |
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75 sip_tx_packet(&msg, &call->udp_sin); |
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76 call->sip_tx_count++; |
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77 } else { |
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78 syslog(LOG_ERR, |
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79 "Call in%06u: late error retrans timeout", |
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80 call->in_tag_num); |
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81 call->sip_state = SIP_STATE_ENDED; |
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82 sip_mark_end_time(call, sip_linger_error); |
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83 transition_dead_sip(call); |
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84 } |
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85 break; |
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86 case SIP_STATE_BYE_SENT: |
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87 if (call->sip_tx_count < cfg_retrans_count) { |
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88 fill_bye_out_msg(&msg, call); |
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89 sip_tx_packet(&msg, &call->udp_sin); |
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90 call->sip_tx_count++; |
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91 } else { |
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92 syslog(LOG_ERR, |
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93 "Call in%06u: BYE-out retrans timeout", |
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94 call->in_tag_num); |
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95 call->sip_state = SIP_STATE_ENDED; |
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96 sip_mark_end_time(call, sip_linger_error); |
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97 transition_dead_sip(call); |
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98 } |
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99 break; |
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100 } |
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101 } |
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102 } |