Hi, I am Aabhas Garg, Collaboration Tech Specialist, working with Professor Gurcharan at Rochester Institute of Technology. Generally, I have gone through this link http://www.pjsip.org/pjsua.htm#config_file on PJSIP website and have developed great interest in your website. Thank you for the information about PJSIP Phone ! Frankly, I will try to find your own source code through PJSIP in addition to video conferencing (vic) and Robust audio Tool (rat) of Access Grid following by python (www.accessgrid.org). As you said, running pjsua through command line, Running pjsua without any arguments will bind pjsua to TCP and UDP port 5060 of local host: *./pjsua* sip:192.168.0.10 and options --add-codec=name ... instead of ./pjsua sip:192.168.0.10 -add-codec=vic (host/port) -audio(host/port) ?? Please let me know if it is possibleto correct on Windows and Linux as I intend to do thesis/or project for master next quarter. Please let me know.. * * ***Could you kindly provide your assistance in this regard ? Thanks and sincerely,* --Aabhas Garg Collaboration Technologist Research Computing Rochester Institute of Technology On Tue, Nov 24, 2009 at 8:08 AM, <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. NAT and one-way voice (fcch2k) > 2. Emergency Call, Call Record and Three Party VOIP PjLib. > (Wong Peter) > 3. Called party preferences Pjlib (Wong Peter) > 4. Re: Called party preferences Pjlib (P.Muge Ersoy) > 5. Re: NAT and one-way voice (RobertT) > 6. Re: About the IPv4/v6 dual stack support (Klaus Darilion) > 7. Re: pjsua_call.c Unable to create media session: Invalid > media payload type (Benny Prijono) > 8. Re: Surviving to IP address changes (Benny Prijono) > 9. Re: Surviving to IP address changes (Saul Ibarra Corretge) > 10. Re: pjsua: no audio is heard (Benny Prijono) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 23 Nov 2009 15:13:08 -0800 > From: fcch2k <fcch2000@xxxxxxxxx> > Subject: NAT and one-way voice > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: > <2ba6b7910911231513p5b2e702am89d0e78d54138472 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > Hi, > > I used pjsip 1.03 and register to SIP server ONLY (without STUN, > without ICE, without TURN), > and also use the SIP server as outbound proxy. > > Sometimes only has one way voice in this configuration, I am wondering > whether the on-way voice is cause by the NAT issue. > > How to enbale NAT or how to know the NAT is used or is enabled in this > configuration? > > Thanks, > > -fcch > > > > ------------------------------ > > Message: 2 > Date: Tue, 24 Nov 2009 11:53:49 +0800 > From: Wong Peter <peterapiit@xxxxxxxxx> > Subject: Emergency Call, Call Record and Three Party VOIP > PjLib. > To: Pjlib Mailing List <pjsip at lists.pjsip.org> > Message-ID: > <dcb7d8ea0911231953g7dce20bi3169c297909725dd at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello to all, i have question regarding the topic above. > > 1. How to make emergency call using softphone and trace emergency origin > voip call from PSTN network ? > 2. Any example of call record and call playback ? > 3. How to three people voip call ? > > Please help. > > I need do my assignment. > > Thanks. > > -- > Linux > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20091124/86d75487/attachment-0001.html > > > > ------------------------------ > > Message: 3 > Date: Tue, 24 Nov 2009 11:55:01 +0800 > From: Wong Peter <peterapiit@xxxxxxxxx> > Subject: Called party preferences Pjlib > To: Pjlib Mailing List <pjsip at lists.pjsip.org> > Message-ID: > <dcb7d8ea0911231955n39b3ecd2p50f5c3be09626917 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hello to all, > > 1. how to implement Called party preferences in pjlib ? Accept call or > reject call. > > Thanks. > > -- > Linux > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20091124/696ecf1e/attachment-0001.html > > > > ------------------------------ > > Message: 4 > Date: Tue, 24 Nov 2009 08:38:00 +0200 > From: "P.Muge Ersoy" <muge.ersoy@xxxxxxxxx> > Subject: Re: Called party preferences Pjlib > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: > <ad39e2590911232238m61f987f6k35f9938d78d49da9 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Please check pjsip-apps\src\pjsua\pjsua_app.c > > On incoming call below callback function is triggering. You may accept or > reject call upon request. > > > app_config.cfg.cb.on_incoming_call = &on_incoming_call; > > Muge > > On Tue, Nov 24, 2009 at 5:55 AM, Wong Peter <peterapiit at gmail.com> wrote: > > > Hello to all, > > > > 1. how to implement Called party preferences in pjlib ? Accept call or > > reject call. > > > > Thanks. > > > > -- > > Linux > > > > _______________________________________________ > > 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/20091124/867f0a80/attachment-0001.html > > > > ------------------------------ > > Message: 5 > Date: Tue, 24 Nov 2009 10:57:32 +0300 > From: RobertT <siniypin@xxxxxxxxx> > Subject: Re: NAT and one-way voice > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: > <2160023e0911232357m45803575j5324c0e6deae4310 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > You should use stun address resolution in order to find out whether you are > behind NAT or not. There is a callback returning discovered NAT type in > pjsua ( > > http://www.pjsip.org/pjsip/docs/html/structpjsua__callback.htm#a992011976963178f5b831d93ee058b7 > ). > > Regards, Robert. > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20091124/8320d0b8/attachment-0001.html > > > > ------------------------------ > > Message: 6 > Date: Tue, 24 Nov 2009 10:06:06 +0100 > From: Klaus Darilion <klaus.mailinglists@xxxxxxxxx> > Subject: Re: About the IPv4/v6 dual stack support > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: <4B0BA1FE.5010208 at pernau.at> > Content-Type: text/plain; charset=GB2312 > > > > c z schrieb: > > Hi, All > > > > I have one question about the pjsip stack: > > > > Is this sip stack supporting IPv4/IPv6 dual-stack receiving > spontaneously? > > I mean whether the Sip messages from IPv4 and IPv6 hosts can be received > > at the same > > time? > > I have not tried it yet, but as IPv4 and v6 are different transports, it > should work if you create 2 transports - one with v4 and one with v6. > > regards > Klaus > > PS: pjsip supports on IPv6 only UDP > > > > > > Thanks very much! > > > > Best regards > > > > Charles Zhang > > > > ------------------------------------------------------------------------ > > ????????????????? > > < > http://cn.rd.yahoo.com/mail_cn/tagline/card/*http://card.mail.cn.yahoo.com/ > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > 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: 7 > Date: Tue, 24 Nov 2009 19:44:04 +0700 > From: Benny Prijono <bennylp@xxxxxxxxx> > Subject: Re: pjsua_call.c Unable to create media session: > Invalid media payload type > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: > <49a4be80911240444v32204644xa5449203d232e994 at mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > On Mon, Nov 23, 2009 at 3:17 PM, Nicholas Poon <nicholas_poon at yahoo.com > >wrote: > > > Hi, many thanks. Attached please find the log for your reference. > > > > Feel free to advise for any hints!! > > > > > There's something wrong with the other end. In the INVITE request, payload > type 113 is offered as iLBC, but remote set payload type 113 as G726-24 in > the answer. So we hung up. > > Cheers > Benny > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20091124/a90956f5/attachment-0001.html > > > > ------------------------------ > > Message: 8 > Date: Tue, 24 Nov 2009 19:53:53 +0700 > From: Benny Prijono <bennylp@xxxxxxxxx> > Subject: Re: Surviving to IP address changes > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: > <49a4be80911240453n7a8f0a89lf557e9b5f9112243 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Sun, Nov 22, 2009 at 6:44 PM, Saul Ibarra Corretge > <saul at ag-projects.com> wrote: > > > > On Nov 21, 2009, at 11:25 AM, oto nag wrote: > > > >> Hi, > >> > >> I am trying to build an mobile sip client (in python for now) and I have > same problems as written on the page you posted. If do call.reinvite() in > SDP there is still an od IP. > >> > >> For IP change detection I made an another thread in witch I monitor > actual IP address (many times per second) and if change is detected I can > execute a proper actions.. but question for me is, what should be proper > actions. And if these actions can be achieved with pjsua. > >> > > > > You'll nedd to restart the transport and the accounts... and if you're > using pjsua I think you'll also need to restart the whole pjsua. > > > > I'm not sure how did you come up with that conclusion. The wiki has > said that you don't need to do that. Or is there other issues that the > wiki haven't covered? > > Cheers > Benny > > > > ------------------------------ > > Message: 9 > Date: Tue, 24 Nov 2009 13:59:33 +0100 > From: Saul Ibarra Corretge <saul@xxxxxxxxxxxxxxx> > Subject: Re: Surviving to IP address changes > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: <C11AD734-C996-44EF-82C8-9955CC8A60E6 at ag-projects.com> > Content-Type: text/plain; charset=us-ascii > > > > > I'm not sure how did you come up with that conclusion. The wiki has > > said that you don't need to do that. Or is there other issues that the > > wiki haven't covered? > > > > Oh, I'm glad to be wrong :) > > So how does PJSIP handle this actually? We are currently not using PJSUA > bindings. The test I made was to launch pjsua console app and then change my > IP. After that I could make a call, but it was going out through the old IP. > Is the contact rewrite (IIRC) option the only way to fix it? > > And fot the media, a transport restart is needed, right? > > I read the wiki page regarding Symbian and IP survival, please point me > elsewhere if that's not the correct place. > > > Regards, > > -- > Saul Ibarra Corretge > AG Projects > > > > > > > > ------------------------------ > > Message: 10 > Date: Tue, 24 Nov 2009 20:08:46 +0700 > From: Benny Prijono <bennylp@xxxxxxxxx> > Subject: Re: pjsua: no audio is heard > To: pjsip list <pjsip at lists.pjsip.org> > Message-ID: > <49a4be80911240508i3a1d19f8u5114788ba071fa32 at mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > On Fri, Nov 20, 2009 at 9:43 PM, Peter Lukac <p.lukac at emtest.sk> wrote: > > oh thanks > > i use this option but without option --playback-dev=1 :) it work now.. > > but i get lot of logs when run pjsua: > > > > ?15:32:50.407 ? ? ec0x257c50 ?Buffer size adjusted from 984 to 745 > (eff_cnt=720) > > ?15:32:50.467 ? ? ec0x257c50 ?Buffer size adjusted from 905 to 666 > (eff_cnt=720) > > > > and when i calling i have lot of logs : > > > > ?15:33:52.130 ? Master/sound ?Buffer size adjusted from 804 to 656 > (eff_cnt=640) > > ?15:33:52.521 ? ? ec0x257c50 ?Underflow, buf_cnt=84, will generate 1 > frame > > ?15:33:52.531 ? Master/sound ?Buffer size adjusted from 816 to 658 > (eff_cnt=640) > > ?15:33:52.746 ? ? ec0x257c50 ?Underflow, buf_cnt=84, will generate 1 > frame > > ?15:33:52.841 ? Master/sound ?Buffer size adjusted from 818 to 645 > (eff_cnt=640) > > > > when i calling pc - pc i have nothing.... > > > > and for speex codec (speex/8000/1) it not works..i have intermittent > sound > > still > > > > Those are probably signs that you're running out of CPU processing > power. Make sure that you build pjsip with the right compilation > settings, e.g. --disable-floating-point configure option, gcc > --mcpu/--march, and so on. Also pjsip needs to be configured to work > optimally on these kind of embedded systems, for a sample config see > the WinCE section on config_site_sample.h (especially > PJ_HAS_FLOATING_POINT, PJMEDIA_CODEC_SPEEX_DEFAULT_QUALITY, and > PJMEDIA_HAS_SPEEX_AEC). > > Cheers > Benny > > > > ------------------------------ > > _______________________________________________ > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > End of pjsip Digest, Vol 27, Issue 66 > ************************************* > -- Thanks and Regards, Aabhas Garg -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20091124/af1b75e9/attachment-0001.html>
