[SIPForum-discussion] Bulk Call Generator

[Jon DiVita]

Sipp is a great free tool for bulk call generation.

Wireshark, ngrep, tcpdump for packet analysis.

As far as commercial products; Empirix Hammer and Ixia traffic generators
are extremely powerful tools.

Hope this helps!

Jon

On Tue, Aug 25, 2009 at 7:06 AM, Manish Jain <jain.manish30 at gmail.com>wrote:

> Dear All,
>
> I am looking for a SIP based Bulk Call generator and Packet analyzer.
>
> Can anybody help me out the name of free OR paid products with the vendor
> name?
>
> Thanks
> Manish Jain
>
> On Mon, Aug 24, 2009 at 10:32 AM, Rachel Baskaran <
> rachelbaskaran at gmail.com> wrote:
>
>> Hey,
>>
>> Now If I already have a ASTERISK at my workplace, how I need to connect to
>> my other SIP acc or say my colleague?
>>
>> Will the following script be enough to do the job?
>>
>> #include<pjsua-lib/pjsua.h>
>>
>> #define THIS_FILE       "APP"
>> #define SIP_DOMAIN   "test.com" ( where my company domain fits here )
>> #define SIP_USER       "rachel"
>> #define SIP_PASSWD   "test"
>>
>> *//function call for incoming call with acc_id, call_id*
>> 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));
>>
>> pjsua_call_answer(call_id, 200, NULL, NULL);
>> }
>>
>>
>> *//main() where argv[] contains URI to call to*
>> int main(int argc, char *argv[])
>> {
>> pjsua_acc_id acc_id;
>> pj_status_t status;
>>
>>
>> status = pjsua_create();
>> if(status != PJ_SUCESS)
>> error_exit("Error in pjsua_create()", status);
>>
>>
>> if(argc > 1) {
>> status = pjsua_verify_sip_url(argv[1]);
>> if(status != PJ_SUCESS)
>> error_exit("Invalid URL in argv", status);
>> }
>>
>> *//Variable dec for config files*
>> pjsua_config cfg;
>> pjsua_logging_config log_cfg;
>>
>> pjsua_config_default(&cfg);
>> cfg.cb.on_incoming_call = &on_incoming_call;
>>
>> pjsua_logging_config_default(&log_cfg);
>> log_cfg.console_level = 4;
>>
>>
>> *//Intilize pjsua *
>> status = pjsua_init(&cfg, &log_cfg, NULL);
>> if(status != PJ_SUCESS)
>> error_exit("Error in pjsua_init()", status);
>> }
>>
>> *//transport through UDP *
>> {
>> pjsua_transport_config cfg;
>>
>>
>> pjsua_transport_config_default(&cfg);
>> cfg.port = 5060;
>>  status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
>> if(status != PJ_SUCESS)
>> error_exit("Error creating transport", status);
>> }
>>
>> status = pjsua_start();
>> if(status != PJ_SUCESS)
>> error_exit("Error starting pjsua", status);
>> }
>>
>> if(argc > 1){
>> pj_str_t uri = pj_str(argv[1]);
>> status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
>> if(status != PJ_SUCESS)
>> error_exit("Error making call", status);
>> }
>>
>> for(;;){
>> char option[10];
>>
>> puts("Press 'h to hangup all calls, 'q'  to quit");
>> if(fgets(option, sizeof(option), stdin) == NULL){
>> puts("EOF while reading stdin, will quit now...");
>> break;
>> }
>>
>> if (option[0] == 'q')
>> break;
>> if(option[0] == 'h')
>> pjsua_call_hangup_all();
>> }
>> pjsua_destroy();
>> return 0;
>> }
>>
>>
>> let's consider for example, that I'm gonna edit my sip.conf file as
>> *[general]
>>
>> [1000] (unique identifier)
>> type= friend ( includes both user and peer )
>> context = phones
>> host = dynamic ( client has a dynamic IP address )*
>>
>>
>> Does the acc_id refers to my id( i.e rachel ) or my colleague's? What
>> about the call_id?
>>
>> If this script is right, where I need to store my script and run with GNU
>> compiler?
>>
>> Any help would be great!
>>
>>
>> Thanks
>> Rachel
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>>
>> _______________________________________________
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>>
>>
>
>
> --
> With Regards,
> Manish Jain
>
> Luck Is Like a Blank Paper,where we writes our Future.
>
> _______________________________________________
> This is the SIP Forum discussion mailing list
> TO UNSUBSCRIBE, or edit your delivery options, please visit
> http://sipforum.org/mailman/listinfo/discussion
> Post to the list at discussion at sipforum.org
>
>
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