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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