I checked the server configuration and I figured out I made a mistake: the
Kamailio server is set up to use TLS.
Since pjsip libraries are already built with SSL, I just changed the
transport setup:
pjsua_transport_config cfg;
pjsua_transport_config_default(&cfg);
cfg.port = 5080;
status = pjsua_transport_create(PJSIP_TRANSPORT_TLS, &cfg, NULL);
and the account configuration:
cfg.id = pj_str("sips:" SIP_USER "@" SIP_URL);
cfg.reg_uri = pj_str("sips:" SIP_URL ";transport=tls");
Nevertheless, I'm still getting the same route error.
2015-01-12 17:25 GMT+01:00 Bill Gardner <billg at wavearts.com>:
> Maybe because you're using port 5061 it requires additional SSL setup?
> Try 5060 perhaps? - Bill
>
>
> On 1/12/2015 11:18 AM, Alberto Bitto wrote:
>
> Hello,
>
> I'm trying to implement a very simple iOS app that uses pjsua. I
> succesfully built and imported the pjsip library in my sample project, have
> a Kamailio remote server up and running and now I'm trying to make a very
> simple call to the server.
>
> Using the code found on this tutorial
> http://www.xianwenchen.com/blog/2014/06/30/how-to-make-an-ios-voip-app-with-pjsip-part-3/,
> I do the following:
>
> 1) create pjsua status:
>
> status = pjsua_create();
>
> 2) init configuration and logging:
>
> pjsua_config 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 log_cfg;
> pjsua_logging_config_default(&log_cfg);
> log_cfg.console_level = 4;
>
> status = pjsua_init(&cfg, &log_cfg, NULL);
>
> 3) add UDP and TCP transport:
>
> pjsua_transport_config cfg;
> pjsua_transport_config_default(&cfg);
> cfg.port = 5080;
>
> status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
>
> (and the same for TCP)
>
> 4) start pjsua:
>
> status = pjsua_start();
>
> 5) register the account on the server:
>
> pjsua_acc_id acc_id;
> pjsua_acc_config cfg;
>
> pjsua_acc_config_default(&cfg);
>
> cfg.id = pj_str("sip:" SIP_USER "@" SIP_URL);
> cfg.reg_uri = pj_str("sip:" SIP_URL);
> cfg.cred_count = 1;
> cfg.cred_info[0].realm = pj_str(SIP_REALM);
> 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);
>
> where SIP_URL is the IP of my Kamailio server and SIP_REALM = "*"
>
> 6) if everything is fine, make the call:
>
> pj_str_t uri = pj_str(SIP_URL ":" SIP_PORT);
> status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL,
> NULL);
>
> where SIP_PORT is the port 5061 on the Kamailio server.
>
> When I run the app and actually make the call, on the log I can see the
> pjsua status change from NULL up to RUNNING and the account successfully
> added to the server; when it tries to make the call, after the sound media
> setup, I get this error:
>
> *Error making call: Missing route set (for tel: URI) (PJSIP_ENOROUTESET)
> [status=171005]*
>
> and pjsua starts to shut everything down.
>
> What am I missing?
>
> Thanks in advance.
>
> Alberto Bitto
>
>
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