Nanang, I think UA needs to handle 408 at the application level, not stack. Certain environments may have restrictions on when and how to retransmit REGISTRATION after getting 408. the application can easily check the response and perform restart REGISTRATION upon receiving 408. BTW, the same will also hold true for INVITE. Alex On Fri, Mar 19, 2010 at 1:00 PM, <pjsip-request at lists.pjsip.org> wrote: > Send pjsip mailing list submissions to > pjsip at lists.pjsip.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > or, via email, send a message with subject or body 'help' to > pjsip-request at lists.pjsip.org > > You can reach the person managing the list at > pjsip-owner at lists.pjsip.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of pjsip digest..." > > > Today's Topics: > > 1. Re: G723 Codec Issues (Nanang Izzuddin) > 2. Re: libpjsua stops reREGISTER after 408 (Nanang Izzuddin) > 3. Re: Mobile Call problem (Nanang Izzuddin) > 4. Re: How to connect beep tone to current call session > (Nanang Izzuddin) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 19 Mar 2010 21:30:01 +0700 > From: Nanang Izzuddin <nanang@xxxxxxxxx> > To: pjsip list <pjsip at lists.pjsip.org> > Subject: Re: G723 Codec Issues > Message-ID: > <f8a01ced1003190730u72e26969r6e182a66566cee02 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi Rams, > > Couldn't tell anything except that the codec parameter for pjmedia > seems to be fine which is important for SDP negotiation and media > framework setup :) > > Perhaps something about G.723 implementation/configuration > incompatibility, such as VAD was used but not really supported in one > side, or extremely/unlikely the PSTN side expected only 5300bps > bitrate. > > Bad RTP message may be harmless as you got audible audio, also not > sure about the usage of static payload type 19 there. > > BR, > nanang > > > On Fri, Mar 12, 2010 at 8:17 PM, rams <rammeth at yahoo.co.in> wrote: > > > > Hi Benny/Nanang, > > > > ? I integrated G723 Codec in to PJSIP successfully, > > > > These are the parameters iam using while configuring G723 to PJSIP > > > > pj_bzero(attr, sizeof(pjmedia_codec_param)); > > ??? attr->info.clock_rate = 8000; > > ??? attr->info.channel_cnt = 1; > > ??? attr->info.avg_bps = 6300; > > ??? attr->info.pcm_bits_per_sample = 16; > > ??? attr->info.frm_ptime = 30; > > ??? attr->info.pt = PJMEDIA_RTP_PT_G723; > > > > ??? attr->setting.frm_per_pkt = 1; > > ??? attr->setting.vad = 1; > > #if !PLC_DISABLED > > ??? attr->setting.plc = 1; > > #endif > > > > /* Enable high pass filter */ > > ??? g723_set_hp(1); > > > > ??? /* Enable post filter */ > > ??? g723_set_pf(1); > > > > ??? /* Enable VAD */ > > ??? g723_set_vx(0); > > > > ??? /* Use 6.3kbps rate */ > > ??? g723_set_rate(0); > > > > > > In X SIP Server i created two Accounts A and B > > > > If i call from A to B voice is audible at both sides with out > disturbance. > > > > while iam calling from A to PSTN Mobile Call,from A side voice is audible > clearly at Mobile side voice with disturbance. > > At the same time after call connected iam getting Bad RTP > > > > ?pjsua_app.c? Media for call 1 is active > > ?sound_port.c? EC activated > > ?strm00D158BC? VAD re-enabled > > ?strm00D158BC? Bad RTP pt 19 (expecting 4) > > ?strm00D158BC? Bad RTP pt 19 (expecting 4) > > ?strm00D158BC? Bad RTP pt 19 (expecting 4) > > ?strm00D158BC? Bad RTP pt 19 (expecting 4) > > ?strm00D158BC? Bad RTP pt 19 (expecting 4) > > > > if i use the GSM codec wt PJSIP is provided,after call connecting > everything fine. > > wt is the problem while adding the G723 codec,could you explain > clearly... > > > > Regards > > Rams > > > > > > > > > > > > > > > > > > > > ________________________________ > > The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. > > _______________________________________________ > > 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 > > > > > > ------------------------------ > > Message: 2 > Date: Fri, 19 Mar 2010 21:58:11 +0700 > From: Nanang Izzuddin <nanang@xxxxxxxxx> > To: pjsip list <pjsip at lists.pjsip.org> > Subject: Re: libpjsua stops reREGISTER after 408 > Message-ID: > <f8a01ced1003190758h463e2fbdk49887e12806b9957 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi Klaus, > > Basically, it is application responsibility to update the registration > state in network up/down events, as currently network state monitoring > should be done in application level and it sounds useless for pjsip to > re-register when network is down. > > Just FYI, ticket #1032 has added TCP/TLS transport state notification > feature, so when SIP transport TCP/TLS reports network disconnection > (based on transport failure in sending keep-alive or any SIP > messages), application can try to initiate/monitor network > reconnection, and application may start re-register once network is > up. > > BR, > nanang > > > On Fri, Mar 12, 2010 at 4:26 PM, Klaus Darilion > <klaus.mailinglists at pernau.at> wrote: > > Hi! > > > > QjSimple client stops registering if one REGISTER times out (e.g. network > > was down). > > > > ?02:34:04.239 ? ?pjsua_acc.c ?sip:user2 at sip.nxdomain.at<sip%3Auser2 at sip.nxdomain.at>: > registration > > success, status=200 (OK), will re-register in 60 seconds > > QjSimple: Info: on_reg_state_slot called > > QjSimple: Info: on_reg_state_slot: status=200 > > QjSimple: Info: on_reg_state_slot: statustext=OK > > ?02:34:59.458 ? ?pjsua_acc.c ?sip:user2 at sip.nxdomain.at<sip%3Auser2 at sip.nxdomain.at>: > registration > > success, status=200 (OK), will re-register in 60 seconds > > QjSimple: Info: on_reg_state_slot called > > QjSimple: Info: on_reg_state_slot: status=200 > > QjSimple: Info: on_reg_state_slot: statustext=OK > > ?02:36:26.458 ? ?pjsua_acc.c ?SIP registration failed, status=408 > (Request > > Timeout) > > QjSimple: Info: on_reg_state_slot called > > QjSimple: Info: on_reg_state_slot: status=408 > > QjSimple: Info: on_reg_state_slot: statustext=Request Timeout > > > > Now, it does not REGISTER again (I waited several hours). > > > > Shouldn't pjsip recover itself and try to REGISTER after a few minutes? > > > > Or is this something I have to handle in the application? > > > > thanks > > Klaus > > > > > > > > _______________________________________________ > > 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 > > > > > > ------------------------------ > > Message: 3 > Date: Fri, 19 Mar 2010 22:51:44 +0700 > From: Nanang Izzuddin <nanang@xxxxxxxxx> > To: pjsip list <pjsip at lists.pjsip.org> > Subject: Re: Mobile Call problem > Message-ID: > <f8a01ced1003190851m2932de3es760c4faedf619c8e at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > > Could you provide the log file and packet capture? > > Shooting in the dark here.. > It seems that you expect the response come via UDP transport, however > the message might be sent via TCP (i.e: when the message is larger > than MTU, by default set to 1300 bytes). If the target didn't actually > listen on TCP, normally pjsip would try to re-send it using UDP. > However, if the transaction timeout occurred before transport failure > report received, the transaction might have got destroyed and no > re-send performed. > > So perhaps you can try to set PJSIP_DONT_SWITCH_TO_TCP to 1, and see > if the call goes fine. > > BR, > nanang > > > On Fri, Mar 12, 2010 at 9:11 PM, P.Muge Ersoy <muge.ersoy at gmail.com> > wrote: > > Hi All and Benny; > > > > I have a problem with mobile pjsip setup.. I can able to register the > client > > but it seems like transportation is stopping :S > > udp_on_read_complete? calls register call backs fine.. But when i try to > > call a number .. It never pass from udp_on_read_complete .. after a while > on > > call state call back called with Disconnected msg from timer..? as you > can > > see from below.. I defined Windows Mobile at config_site.h..? Do you have > > any idea why calling is not working in a weird way .. > > > > sip.exe!on_call_state(int call_id = 0, pjsip_event* e = 0x2841e0e4) Line: > > 1046, Byte Offsets: 0x88??? C++ > > ???? sip.exe!pjsua_call_on_state_changed(pjsip_inv_session* inv = > > 0x005499c0, pjsip_event* e = 0x2841e0e4) Line: 3080, Byte Offsets: 0x648 > > C > > ???? sip.exe!inv_set_state(pjsip_inv_session* inv = 0x005499c0, > > pjsip_inv_state state = 6, pjsip_event* e = 0x2841e0e4) Line: 227, Byte > > Offsets: 0x110??? C > > ???? sip.exe!inv_on_state_calling(pjsip_inv_session* inv = 0x005499c0, > > pjsip_event* e = 0x2841e0e4) Line: 3259, Byte Offsets: 0x2a4??? C > > ???? sip.exe!mod_inv_on_tsx_state(pjsip_transaction* tsx = 0x0054ad54, > > pjsip_event* e = 0x2841e0e4) Line: 565, Byte Offsets: 0x88??? C > > ???? sip.exe!pjsip_dlg_on_tsx_state(pjsip_dialog* dlg = 0x00547944, > > pjsip_transaction* tsx = 0x0054ad54, pjsip_event* e = 0x2841e0e4) Line: > > 1908, Byte Offsets: 0xf0??? C++ > > ???? sip.exe!mod_ua_on_tsx_state(pjsip_transaction* tsx = 0x0054ad54, > > pjsip_event* e = 0x2841e0e4) Line: 180, Byte Offsets: 0x54??? C > > ???? sip.exe!tsx_set_state(pjsip_transaction* tsx = 0x0054ad54, > > pjsip_tsx_state_e state = 6, pjsip_event_id_e event_src_type = 1, void* > > event_src = 0x0054ae58) Line: 1116, Byte Offsets: 0x1fc??? C > >>??? sip.exe!tsx_on_state_calling(pjsip_transaction* tsx = 0x0054ad54, > >> pjsip_event* event = 0x2841e160) Line: 2166, Byte Offsets: 0x128??? C > > ???? sip.exe!tsx_timer_callback(pj_timer_heap_t* theap = 0x00430230, > > pj_timer_entry* entry = 0x0054ae58) Line: 1057, Byte Offsets: 0xb4??? C > > ???? sip.exe!pj_timer_heap_poll(pj_timer_heap_t* ht = 0x00430230, > > pj_time_val* next_delay = 0x2841e1c8) Line: 519, Byte Offsets: 0x17c??? C > > > > > > _______________________________________________ > > 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 > > > > > > > > ------------------------------ > > Message: 4 > Date: Fri, 19 Mar 2010 22:54:58 +0700 > From: Nanang Izzuddin <nanang@xxxxxxxxx> > To: pjsip list <pjsip at lists.pjsip.org> > Subject: Re: How to connect beep tone to current call session > Message-ID: > <f8a01ced1003190854g6c6ae1ces3c1550158dd36b7e at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi Bharat, > > Seems that you are using APS/VAS-Direct. As mentioned in the > APS/VAS-Direct wiki, audio switch cannot mix audio, so sound port can > only listen to one port at a time. In case the sound port is > busy/connected, the behaviour of pjsua_conf_connect(x, 0) will be: > - if the audio param (format, bitrate, clockrate, etc) of port x > matches to sound port, current sound port connection will be released > and the new connection (x -> 0) created > - otherwise, the connection request will be rejected. > > So try to release any sound device connection in the audio switch > first before connecting the beep port, and later do the same thing to > return back the call stream port. This will disrupt the audio call as > sound device will be reopened twice and incoming RTP packets are not > consumed during the beep. > > BR, > nanang > > > On Fri, Mar 19, 2010 at 7:03 PM, Bharat Yadav > <bharat.yadav at axisconvergence.com> wrote: > > Hi Nanang > > I am using Symbian and trying to play beep tone as minute minder but when > i > > went to connect beep port using "status = > > pjsua_conf_connect(app_config.beep_slot, 0);", it played well when no > voip > > call, if any active call no status shows only 220160 or 171140, whats > wrong, > > please guide me. > > Thanks and regards > > Bharat > > > > On Fri, Mar 19, 2010 at 4:29 PM, Nanang Izzuddin <nanang at pjsip.org> > wrote: > >> Hi Bharat, > >> > >> Not really sure what is the problem as you can play dtmf tones > >> already, now it's just that the trigger is a timer, see > >> pjsua_schedule_timer() perhaps? :) > >> > >> BR, > >> nanang > >> > >> > >> On Tue, Mar 9, 2010 at 4:00 PM, Bharat Yadav > >> <bharat.yadav at axisconvergence.com> wrote: > >>> Hi all > >>> > >>> I want to play beep tone to current call session how to achieve it. I > >>> am able to play dtmf tones, but i requires beep tone for minute > >>> minder. > >>> > >>> Thanks > >>> > >>> Bharat > >>> > >>> _______________________________________________ > >>> 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 > >>> > >> > >> _______________________________________________ > >> 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 > >> > > > > _______________________________________________ > > 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 > > > > > > ------------------------------ > > _______________________________________________ > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > End of pjsip Digest, Vol 31, Issue 35 > ************************************* > -------------- next part -------------- An HTML attachment was scrubbed... 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