Check again the log, see if there was something suspicious, or check
connection/firewall things (as some endpoints may consider to close a
call session when receiving no RTP packets). Or why don't you ask the
sender of BYE? ;)
Yes, symbian_ua_gui should be able to.
BR,
nanang
2010/1/27 Bo Song <songbo_highlight at hotmail.com>:
> Hi,
> ?? I build the symbian_ua_gui example, and install it in my Nokia N73
> device. I want to call the PSTN landline. But I'm not sure it will be
> sucessful.
> ?? The symbian_ua_gui setting as:
> ?//config_size define
> ?config_site.h include the pj/config_site_sample.h
> ?........................
> ?//the macro in symbian_ua.cpp
> ?#define SIP_PORT?5060
> ?#define USE_ICE??0
> ?#define USE_SRTP?PJSUA_DEFAULT_USE_SRTP
> ?? ?....................
> ?//the media define
> ??????? pjsua_media_config_default(&med_cfg);
> ??? ?med_cfg.thread_cnt = 0;
> ??? ?med_cfg.has_ioqueue = PJ_FALSE;
> ??? ?med_cfg.clock_rate = 8000;
> ?med_cfg.audio_frame_ptime = 40;
> ??? ?med_cfg.ec_tail_len = 0;
> ??? ?med_cfg.enable_ice = USE_ICE;
> ??? ?med_cfg.snd_auto_close_time = 5;
> ?....................
> ?//the codec
> ??????? pj_str_t codec_id = pj_str("speex/8000");
> ??????? pjmedia_codec_mgr_set_codec_priority(
> ??????? ?pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
> ??????? ?&codec_id, PJMEDIA_CODEC_PRIO_NORMAL+1);
> ??????? codec_id = pj_str("speex/16000");
> ??????? pjmedia_codec_mgr_set_codec_priority(
> ??????? ?pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
> ??????? ?&codec_id, PJMEDIA_CODEC_PRIO_DISABLED);
> ??????? codec_id = pj_str("speex/32000");
> ??????? pjmedia_codec_mgr_set_codec_priority(
> ??????? ?pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt),
> ??????? ?&codec_id, PJMEDIA_CODEC_PRIO_DISABLED);
>
>
> ?The above is my setting. When I make a call to a PSTN landline, the
> landline phone can ring. But if callee confirmed this calling, the caller
> will receive the BYE request, which will make this call to be disconnected.
>
> ?I check the codec method. Mine is:
> ??v=0
> ??o=VoipSwitch 8972 8972 IN IP4 210.52.252.108
> ??s=VoipSIP
> ??i=Audio Session
> ??c=IN IP4 210.52.252.108
> ??t=0 0
> ??m=audio 7972 RTP/AVP 3 101
> ??a=rtpmap:3 GSM/8000
> ??a=rtpmap:101 telephone-event/8000
> ??a=fmtp:101 0-15
> ??a=sendrecv
> ????? But I learn that the PSTN landline codec is G.711, G.723, G.729 and so
> on.
> ????? So I'm doult that the different codec between pjsip and PSTN results
> of the BYE request from the callee.
>
> ????? I want to ask :
> ?1.The reason why the caller receive BYE request after the callee confirmed
> the calling.
> ?2.Can I make a call to PSTN landline via symbian_ua_gui?
> ??? Thanks!
>
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