Re: [Opalvoip-devel] Opalvoip-devel Digest, Vol 4, Issue 25
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Re: [Opalvoip-devel] Opalvoip-devel Digest, Vol 4, Issue 25
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From: Towhid I. <to...@in...> - 2008-01-26 08:30:42
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*hi read this ->http://www.voxgratia.org/docs/faq.html#7_1 10.4 - How do I use SimpleOPAL to make a call? * you can make call by simpleopal -n sip:192.168.1.247 192.168.150.247 is the ip of your pc that your are calling towhid * * On 1/26/08, opa...@li... < opa...@li...> wrote: > > Send Opalvoip-devel mailing list submissions to > opa...@li... > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.sourceforge.net/lists/listinfo/opalvoip-devel > or, via email, send a message with subject or body 'help' to > opa...@li... > > You can reach the person managing the list at > opa...@li... > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Opalvoip-devel digest..." > > > Today's Topics: > > 1. Help in University Project ( Ayse ?zdemir ) > 2. Re: Help in University Project (Derek Smithies) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 26 Jan 2008 01:39:39 +0200 > From: " Ayse ?zdemir " <ays...@gm...> > Subject: [Opalvoip-devel] Help in University Project > To: opa...@li... > Message-ID: > <603...@ma...> > Content-Type: text/plain; charset="iso-8859-1" > > Hello dear list members, > > I'm a student in a university, and my graduation project is on SIP > implementation. I find out OPAL last week, but I coldn't use it. I've > installed it from tarball with PTLIB. However, I cannot start a call to a > SIP user. Here is the messages when I try to make a call: > > ./simpleopal sip:100@192.168.2.23 -u 123 -p 123 -a -r 123 --sip-domain > 192.168.2.23 --sip-proxy sip:333:333@192.168.2.23 > SimpleOPAL Version 3.0.1 by Open Phone Abstraction Library on Unix Linux ( > 2.6.22.9-91-x86_64) > > 0:00.014 SimpleOPAL (0) Version > 3.0.1 by Open Phone Abstraction Library on Unix Linux (2.6.22.9-91-x86_64) > at 2008/1/26 1:33:51.326 > Jitter buffer: 50-250 ms > TCP ports: 0-0 > UDP ports: 0-0 > RTP ports: 5000-5999 > RTP IP TOS: 0x10 > STUN server: None > Auto answer is on > Sound output device: "" > Sound input device: "" > Video output device: "SDL" > Video input device: "shm" > h323 Local username: 123 > h323 FastConnect is on > h323 H245Tunnelling is on > h323 gk Token OID is > h323 listeners: tcp$*:1720 > Gatekeeper: none. > Could not register with gatekeeper. > h323s Local username: 123 > h323s FastConnect is on > h323s H245Tunnelling is on > h323s gk Token OID is > h323s listeners: tcps$*:1300 > Gatekeeper: none. > Could not register with gatekeeper. > SIP started on udp$*:5060,tcp$*:5060 > Using SIP registrar 123 ... done. > Local endpoint type: pc:* > Codecs removed: > Codec order: G.723.1,G.729B,G.729A/B,G.729,G.729A,GSM-06.10,G.728, > G.711-uLaw-64k,G.711-ALaw-64k > Available codecs: G.711-uLaw-64k,G.711-ALaw-64k > ,RFC4175_YCbCr-4:2:0,RFC4175_RGB > 0:20.043 SimpleOPAL main.cxx(1256) Simple Registered > media formats: > Format Name = YUV420P > Session ID = 2 Video > Payload Type = [pt=127] > Encoding Name = > Adaptive Packet Delay (R/W) = 0 > Clock Rate (R/O) = 90000 > Dynamic Video Quality (R/W) = 0 > Encoding Quality (R/W) = 15 > Frame Height (R/W) = 1152 > Frame Time (R/W) = 3000 > Frame Width (R/W) = 1408 > Max Bit Rate (R/W) = 583925760 > Target Bit Rate (R/W) = 10000000 > > Format Name = RFC4175_YCbCr-4:2:0 > Session ID = 2 Video > Payload Type = [pt=97] > Encoding Name = raw > Adaptive Packet Delay (R/W) = 0 > Clock Rate (R/O) = 90000 FMTP name: rate () > Dynamic Video Quality (R/W) = 0 > Encoding Quality (R/W) = 15 > Frame Height (R/W) = 1080 FMTP name: height () > Frame Time (R/W) = 1500 > Frame Width (R/W) = 1920 FMTP name: width () > Max Bit Rate (R/W) = 186624000 > rfc4175_colorimetry (R/O) = "BT601-5" FMTP name: colorimetry () > rfc4175_depth (R/O) = 8 FMTP name: depth () > rfc4175_sampling (R/O) = "YCbCr-4:2:0" FMTP name: sampling () > Target Bit Rate (R/W) = 10000000 > > Format Name = RGB24 > Session ID = 2 Video > Payload Type = [pt=127] > Encoding Name = > Adaptive Packet Delay (R/W) = 0 > Clock Rate (R/O) = 90000 > Dynamic Video Quality (R/W) = 0 > Encoding Quality (R/W) = 15 > Frame Height (R/W) = 1152 > Frame Time (R/W) = 3000 > Frame Width (R/W) = 1408 > Max Bit Rate (R/W) = 1167851520 > Target Bit Rate (R/W) = 10000000 > > Format Name = RFC4175_RGB > Session ID = 2 Video > Payload Type = [pt=96] > Encoding Name = raw > Adaptive Packet Delay (R/W) = 0 > Clock Rate (R/O) = 90000 FMTP name: rate () > Dynamic Video Quality (R/W) = 0 > Encoding Quality (R/W) = 15 > Frame Height (R/W) = 1080 FMTP name: height () > Frame Time (R/W) = 1500 > Frame Width (R/W) = 1920 FMTP name: width () > Max Bit Rate (R/W) = 373248000 > rfc4175_colorimetry (R/O) = "BT601-5" FMTP name: colorimetry () > rfc4175_depth (R/O) = 8 FMTP name: depth () > rfc4175_sampling (R/O) = "RGB" FMTP name: sampling () > Target Bit Rate (R/W) = 10000000 > > Format Name = PCM-16 > Session ID = 1 Audio > Payload Type = [pt=127] > Encoding Name = > Clock Rate (R/O) = 8000 > Frame Time (R/O) = 8 > Max Bit Rate (R/O) = 128000 > Max Frame Size (R/O) = 16 > Needs Jitter (R/O) = 1 > Rx Frames Per Packet (R/W) = 240 > Tx Frames Per Packet (R/W) = 1 > > Format Name = G.711-uLaw-64k > Session ID = 1 Audio > Payload Type = PCMU > Encoding Name = PCMU > Clock Rate (R/O) = 8000 > Frame Time (R/O) = 8 > Max Bit Rate (R/O) = 64000 > Max Frame Size (R/O) = 8 > Needs Jitter (R/O) = 1 > Rx Frames Per Packet (R/W) = 240 > Tx Frames Per Packet (R/W) = 30 > > Format Name = G.711-ALaw-64k > Session ID = 1 Audio > Payload Type = PCMA > Encoding Name = PCMA > Clock Rate (R/O) = 8000 > Frame Time (R/O) = 8 > Max Bit Rate (R/O) = 64000 > Max Frame Size (R/O) = 8 > Needs Jitter (R/O) = 1 > Rx Frames Per Packet (R/W) = 240 > Tx Frames Per Packet (R/W) = 30 > > Format Name = PCM-16-16kHz > Session ID = 1 Audio > Payload Type = [pt=127] > Encoding Name = > Clock Rate (R/O) = 16000 > Frame Time (R/O) = 8 > Max Bit Rate (R/O) = 128000 > Max Frame Size (R/O) = 16 > Needs Jitter (R/O) = 1 > Rx Frames Per Packet (R/W) = 240 > Tx Frames Per Packet (R/W) = 1 > > Format Name = RGB32 > Session ID = 2 Video > Payload Type = [pt=127] > Encoding Name = > Adaptive Packet Delay (R/W) = 0 > Clock Rate (R/O) = 90000 > Dynamic Video Quality (R/W) = 0 > Encoding Quality (R/W) = 15 > Frame Height (R/W) = 1152 > Frame Time (R/W) = 3000 > Frame Width (R/W) = 1408 > Max Bit Rate (R/W) = 1557135360 > Target Bit Rate (R/W) = 10000000 > > Format Name = UserInput/RFC2833 > Session ID = 0 > Payload Type = [pt=101] > Encoding Name = telephone-event > Clock Rate (R/O) = 8000 > Frame Time (R/O) = 1200 > Max Bit Rate (R/O) = 640 > Max Frame Size (R/O) = 4 > Needs Jitter (R/O) = 1 > > Format Name = NamedSignalEvent > Session ID = 0 > Payload Type = [pt=100] > Encoding Name = NSE > Clock Rate (R/O) = 8000 > Frame Time (R/O) = 1200 > Max Bit Rate (R/O) = 640 > Max Frame Size (R/O) = 4 > Needs Jitter (R/O) = 1 > > > Pausing to allow registration to occur...done > Initiating call to "sip:100@192.168.2.23" > Press ? for help. > Command ? Call with "sip:100@192.168.2.23" completed, on Sat 1:34AM. > Duration 0:00s. > -------------- next part -------------- > An HTML attachment was scrubbed... > > ------------------------------ > > Message: 2 > Date: Sat, 26 Jan 2008 21:11:02 +1300 (NZDT) > From: Derek Smithies <de...@in...> > Subject: Re: [Opalvoip-devel] Help in University Project > To: Ayse ?zdemir <ays...@gm...> > Cc: opa...@li... > Message-ID: > <Pin...@ka...> > Content-Type: text/plain; charset="utf-8" > > Hi, > > On Sat, 26 Jan 2008, Ayse ?zdemir wrote: > > > Hello dear list members, > > > > I'm a student in a university, and my graduation project is on SIP > > implementation. I find out OPAL last week, but I coldn't use it. I've > > installed it from tarball with PTLIB. However, I cannot start a call to > a > > SIP user. Here is the messages when I try to make a call: > > > ....Lots deleted. > > Ekiga is based on ptlib/opal and does SIP. > You might like to start here and get ekiga to make a SIP call. > > second > You might like to go to the www.voxgratia.org web site and read some docs > there. > > third > You might like to go to http://www.ekiga.org/ (home page of ekiga) and > read the guides there. In particular, check the FAQ at > http://www.ekiga.org/index.php?rub=3 > > fourth, > It is a bit challenging, and the whole telephony thing can be a bit > difficult at first. However, I see you are a student at university, close > to graduation (this is encouraging - you clearly have the ability to work > things out for yourself). > > Good luck. If you come up with something that is helpful to the project, > please post it back to the list, or offline to one of the key project > members. > > Derek. > > > -- > Derek Smithies Ph.D. > IndraNet Technologies Ltd. > Email: de...@in... > ph +64 3 365 6485 > Web: http://www.indranet-technologies.com/ > > ------------------------------ > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > > ------------------------------ > > _______________________________________________ > Opalvoip-devel mailing list > Opa...@li... > https://lists.sourceforge.net/lists/listinfo/opalvoip-devel > > > End of Opalvoip-devel Digest, Vol 4, Issue 25 > ********************************************* > |