https://stackoverflow.com/questions/23103432
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RussianSince you mention that you've tried the windows phone telnet app sample, I assume you've downloaded the PJSIP winphone source as mentioned in their wp8 getting started guide. To create a simple app that perform outgoing and receive incoming call as you mentioned, you can simply reuse this winphone project.
Open the winphone project and do:
- Create new Windows Phone project and set this as startup project (let's call this project SIP_UI). This will serve as the UI. You can just create a "Call button" that will perform outgoing call later.
- Follow the existing pjsua_wp WMAppManifest.xml settings for this SIP_UI. Especially the capabilities part. Your app won't work if you simply use the default settings.
- Create new Windows Phone Runtime project (let's call this SIP_WINPRT). Create a class and a method inside of this class that will perform the actual call later.
- Change property setting for SIP_WINPRT (right click SIP_WINPRT project -> property) by following the existing pjsua_wp_backend's. Change especially on the References, Additional include directories, and preprocessor definitions. Adjust the path accordingly.
Search for simple_pjsua.c in the winphone sample. And try to implement that in the class you've created in SIP_WINPRT. Sample that I've created:
#include "pch.h" #include "backend.h" #include "pjsua.h" #define SIP_DOMAIN "dogdomain" #define SIP_USER "dog" #define SIP_PASSWD "dog" using namespace backend; using namespace Platform; SipletRuntimeComponent::SipletRuntimeComponent() { } /* Display error and exit application */ static void error_exit(const char *title, pj_status_t status) { //pjsua_perror(THIS_FILE, title, status); pjsua_destroy(); exit(1); } /* Callback called by the library upon receiving incoming call */ static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id, pjsip_rx_data *rdata) { pjsua_call_info ci; PJ_UNUSED_ARG(acc_id); PJ_UNUSED_ARG(rdata); pjsua_call_get_info(call_id, &ci); //PJ_LOG(3,(THIS_FILE, "Incoming call from %.*s!!", // (int)ci.remote_info.slen, // ci.remote_info.ptr)); /* Automatically answer incoming calls with 200/OK */ pjsua_call_answer(call_id, 200, NULL, NULL); } /* Callback called by the library when call's media state has changed */ static void on_call_media_state(pjsua_call_id call_id) { pjsua_call_info ci; pjsua_call_get_info(call_id, &ci); if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) { // When media is active, connect call to sound device. pjsua_conf_connect(ci.conf_slot, 0); pjsua_conf_connect(0, ci.conf_slot); } } /* Callback called by the library when call's state has changed */ static void on_call_state(pjsua_call_id call_id, pjsip_event *e) { pjsua_call_info ci; PJ_UNUSED_ARG(e); pjsua_call_get_info(call_id, &ci); //PJ_LOG(3,(THIS_FILE, "Call %d state=%.*s", call_id, // (int)ci.state_text.slen, // ci.state_text.ptr)); } int SipletRuntimeComponent::SipCall(int address) { /* Create pjsua */ pj_status_t status; status = pjsua_create(); if (status != PJ_SUCCESS){ //Error in pjsua_create() return -1; } /* Validate the URL*/ char url[50] = "sip:cat:cat@catdomain:5060"; status = pjsua_verify_url(url); if (status != PJ_SUCCESS){ //Invalid URL given return -1; } /* Init pjsua */ { pjsua_config cfg; pjsua_logging_config log_cfg; pjsua_config_default(&cfg); cfg.cb.on_incoming_call = &on_incoming_call; cfg.cb.on_call_media_state = &on_call_media_state; cfg.cb.on_call_state = &on_call_state; pjsua_logging_config_default(&log_cfg); log_cfg.console_level = 4; status = pjsua_init(&cfg, &log_cfg, NULL); if (status != PJ_SUCCESS){ //Error in pjsua_init() pjsua_destroy(); return -1; } } /* Add UDP transport. */ { pjsua_transport_config cfg; pjsua_transport_config_default(&cfg); cfg.port = 5060; status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL); if (status != PJ_SUCCESS){ //Error creating transport pjsua_destroy(); return -1; } } /* Initialization is done, now start pjsua */ status = pjsua_start(); if (status != PJ_SUCCESS){ //Error starting pjsua pjsua_destroy(); return -1; } /* Register to SIP server by creating SIP account. */ pjsua_acc_id acc_id; { pjsua_acc_config cfg; pjsua_acc_config_default(&cfg); cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN); cfg.reg_uri = pj_str("sip:" SIP_DOMAIN); cfg.cred_count = 1; cfg.cred_info[0].realm = pj_str(SIP_DOMAIN); cfg.cred_info[0].scheme = pj_str("digest"); cfg.cred_info[0].username = pj_str(SIP_USER); cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD; cfg.cred_info[0].data = pj_str(SIP_PASSWD); status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id); if (status != PJ_SUCCESS){ //Error adding account pjsua_destroy(); return -1; } } /* make call to the URL. */ pj_str_t uri = pj_str(url); status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL); if (status != PJ_SUCCESS){ //Error making call pjsua_destroy(); return -1; } return address + 1; }Add SIP_WINPRT as a reference in SIP_UI project.
- Call the SIP_WINPRT when the Call button is pressed.
