Catually I don't know how to run those examples. I tried building the projects by invoking the ./configure in cygwin, but it returns an error. Attached is the config.log that contains the error. I am developing an application under Symbian Phone N80, I read some of the articles in www.pjsip.org and If my understanding is correct, PJSIP doesn't support SDP. >From sip_transport_udp.c status = pj_sock_setsockopt(sock, pj_SOL_SOCKET(), pj_SO_RCVBUF(), ??? ??? ??? ??? &sobuf_size, sizeof(sobuf_size)); status is always not equal to PJ_SUCCESS and it returns an error sip_transport_ error setting SO_RCVBUF INVALID OPERATION PJ_EINALIDOP sip_transport_ error setting SO_SNDBUF INVALID OPERATION PJ_EINALIDOP Is it possible to resolve this problem? How? >>>>>>>My guess is probably you're missing the call to pjsua_conf_connect(). Actually im using the sample callback functions eg. on_incoming_call(),on_pager etc.. I copied these callbacks from ua.cpp under pjsip-apps/src/symbian_ua. Is these functions ready for voice transmission or do I have to add what you have just stated above. If so, How do I used that function? Sorry but Im new to this and I don't really know what to do, or where to start. A sample code that works might be a good thing to start. Thanks Joseph --- On Tue, 7/29/08, Benny Prijono <bennylp at pjsip.org> wrote: From: Benny Prijono <bennylp@xxxxxxxxx> Subject: Re: HELP!! Voice Connection During A SIP Call To: "pjsip list" <pjsip at lists.pjsip.org> Date: Tuesday, July 29, 2008, 3:52 AM On Tue, Jul 29, 2008 at 1:47 AM, Joseph Maiquez <josephmaiquez at yahoo.com> wrote: My problem was no voice was transmitted during a phone call that's why I can't hear the voice of the person on the other network. I think someone had faced this problem before here is the link.. http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/2008-June/003452.html I think that discussed different problem. ? Im sorry but i dont have any idea on how to transmit voice. ? This is what I did. My guess is probably you're missing the call to pjsua_conf_connect(). ? Call the pjsua_call_make_call function to begin the invite Sip Call Then on the other end, call the function pjsua_call_answer then the call is successfully established but no audio is enable. How can I enable the audio during the call? this is what I wanted to know.. Hope you can help me guys, a sample code on how to make it will be very useful.. But we already have the samples (simple_pjsua, etc.). What's wrong with them? ?-benny _______________________________________________ Visit our blog: http://blog.pjsip.org pjsip mailing list pjsip at lists.pjsip.org http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080729/5ecf3ab9/attachment.html
