GB_Record_Node * GB_Find_Record_Node_by_Call_ID(GB_CONNECT_STATE *gb_cons, osip_call_id_t *call_id, int *index) { GB_Record_Node *record = NULL; int pos; int list_size = 0; if(gb_cons == NULL || call_id == NULL) { TRACE(SCI_TRACE_NORMAL,MOD_GB,"%s line=%d Err\n",__FUNCTION__,__LINE__); return NULL; } list_size = osip_list_size(&(gb_cons->record_node_list)); if(list_size <= 0) { return NULL; } for(pos=0; pos<list_size; pos++) { record = osip_list_get(&(gb_cons->record_node_list), pos); if(record) { if(osip_call_id_match(record->call_id, call_id) == 0) { *index = pos; return record; } record = NULL; } } return NULL; }
static SalOp * sal_find_in_subscribe_by_call_id(Sal *sal, osip_call_id_t *call_id){ const MSList *elem; SalOp *op; for(elem=sal->in_subscribes;elem!=NULL;elem=elem->next){ op=(SalOp*)elem->data; if (op->call_id && osip_call_id_match(op->call_id,call_id)==0) return op; } return NULL; }
static struct session *sip_callback(struct session *sip, u_char *data, uint32_t len) { osip_message_t *msg; struct tuple4 addr; struct session *start; struct session *rtp; osip_call_id_t *call_id; osip_message_init(&msg); if (!osip_message_parse(msg, (char *)data, len)) { if (NULL == sip->u.sip_params.call_id) { /* * If the session object was created by udp_callback * we need to fill in the call_id field here because * udp_callback doesn't know anything about SIP */ if (NULL != (call_id = osip_message_get_call_id(msg))) osip_call_id_clone(call_id, &sip->u.sip_params.call_id); } else { /* * Otherwise check if the session object passed to this * function call was really the one corresponding to the * call ID in the SIP packet, in case several SIP calls * are passed upon the same transport layer protocol, * source and destination IPv4 address & port combination. * udp_callback has no way of knowing how to distinguish * SIP session objects based on call ID, so we have to do * it here. We just continue searching in the list of all * tracked sessions for similar SIP objects until we find * one that has the same call ID, or else we create a new * SIP session object that corresponds to the new call. */ start = sip; do { if (NULL == (call_id = osip_message_get_call_id(msg))) continue; if (!osip_call_id_match(sip->u.sip_params.call_id, call_id)) break; } while ((sip = sessions_find(sip->next, TYPE_SIP, 0, &sip->addr))); if (NULL == sip) { if (bonus_time) { osip_message_free(msg); return start; } sip = sessions_add(start->type, &start->addr, NULL); if (NULL != (call_id = osip_message_get_call_id(msg))) osip_call_id_clone(call_id, &sip->u.sip_params.call_id); } } /* * If the current SIP packet is an INVITE message, store the * advertised source port and IPv4 address. It is not very * important, since we can do only with the destination part * (useful in case the capture missed the INVITE packet), but * it helps discriminating from unrelated packets. * * Unfortunately, some SIP implementations such as the one in * Audiocodes Mediant 1000 SIP gateways actually use a source * port different from the one they advertised in the INVITE * message parameters - how outrageous! - so we have to make * our sessions search engine ignore the source port part by * zeroing it :-/ */ if (MSG_IS_INVITE(msg)) { if (!bonus_time) { sip_get_address(msg, &sip->u.sip_params.rtp_addr.saddr, &sip->u.sip_params.rtp_addr.source); #ifndef USING_NON_STUPID_SIP_IMPLEMENTATIONS sip->u.sip_params.rtp_addr.source = 0; #endif } } else if (MSG_TEST_CODE(msg, 200)) { if (MSG_IS_RESPONSE_FOR(msg, "INVITE")) { if (!bonus_time && sip_get_address(msg, &sip->u.sip_params.rtp_addr.daddr, &sip->u.sip_params.rtp_addr.dest)) { sessions_add(TYPE_UDP | TYPE_RTP, &sip->u.sip_params.rtp_addr, sip); sip->u.sip_params.picked_up = 1; } } else if (MSG_IS_RESPONSE_FOR(msg, "BYE") || MSG_IS_RESPONSE_FOR(msg, "CANCEL")) { start = first_session; while (NULL != (rtp = sessions_find(start, TYPE_RTP, sip->id, NULL))) { sessions_del(rtp); start = rtp->next; } /* * Mark for deletion in 2 seconds, in order to give some * time to the extra ACK packets that might be exchanged */ if (sip->type & TYPE_UDP) sip->timeout = nids_last_pcap_header->ts.tv_sec + 2; } } } osip_message_free(msg); return sip; }