From armangungor at hotmail.com Tue Jan 12 00:02:15 2010 From: armangungor at hotmail.com (Arman Gungor) Date: Tue, 12 Jan 2010 08:02:15 +0000 Subject: [RTPproxy Users] notification on both sides start sending their RTPs? Message-ID: Hello, I have two questions about RTPproxy. Firstly, I want to use Send RTPproxy for communicating 2 phones behind NAT. In order to realize this RTPproxy should wait for both sides to send their RTPs to RTPproxy. But it starts forwarding these RTPs to the other side immediately after receiving from any side. Is there any way to do this without giving a fake IP address in Update message? ( I want to have a control on IPs by forcing ) Secondly, I want to know whether these phones start communicating or not. Is it possible for RTPproxy to send notification when peers start talking successfully (RTPs are being forwarded from both sides) ? Thanks all, Arman Gungor _________________________________________________________________ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail?. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.rtpproxy.org/pipermail/users/attachments/20100112/b5ee0f62/attachment.html From sobomax at sippysoft.com Tue Jan 12 01:57:28 2010 From: sobomax at sippysoft.com (Maxim Sobolev) Date: Tue, 12 Jan 2010 01:57:28 -0800 Subject: [RTPproxy Users] notification on both sides start sending their RTPs? In-Reply-To: References: Message-ID: <4B4C4788.2070407@sippysoft.com> Arman Gungor wrote: > Hello, > > I have two questions about RTPproxy. > > Firstly, I want to use Send RTPproxy for communicating 2 > phones behind NAT. In order to realize this RTPproxy should wait for > both sides to send their RTPs to RTPproxy. But it starts forwarding > these RTPs to the other side immediately after receiving from any side. > Is there any way to do this without giving a fake IP address in Update > message? ( I want to have a control on IPs by forcing ) I am not sure why this scenario is a problem. The proxy will update with correct IP address once it gets the first packet from the other party. Can you please elaborate? > Secondly, I want to know whether these phones start communicating or > not. Is it possible for RTPproxy to send notification when peers start > talking successfully (RTPs are being forwarded from both sides) ? There is no such functionality currently, but it should not be difficult to implement something along the lines. Regards, -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts T/F: +1-646-651-1110 Web: http://www.sippysoft.com MSN: sales at sippysoft.com Skype: SippySoft From armangungor at hotmail.com Tue Jan 12 03:23:09 2010 From: armangungor at hotmail.com (Arman Gungor) Date: Tue, 12 Jan 2010 11:23:09 +0000 Subject: [RTPproxy Users] notification on both sides start sending their RTPs? In-Reply-To: <4B4C4788.2070407@sippysoft.com> References: , <4B4C4788.2070407@sippysoft.com> Message-ID: Imagine the following scenerio, A and B are phones behind NAT. A---------------------NAT-------------------------RTPproxy-----------------------NAT-------------------------------B RTP----------------> -------------------> ------------------> BLOCKED If A sends RTP packets and RTPproxy forwards them before waiting for B's RTPs to come, B's NAT device will block that RTP port, then they would not be able talk at all. Also I want to have a control on IPs of A and B for security reasons. Another person( C ) should not be able send packets to the port reserved for B. If C can be able to send packets to this reserved port then he will be able to hear A's voice. Am I right? Thanks, Arman Gungor > Date: Tue, 12 Jan 2010 01:57:28 -0800 > From: sobomax at sippysoft.com > To: users at rtpproxy.org > Subject: Re: [RTPproxy Users] notification on both sides start sending their RTPs? > > Arman Gungor wrote: > > Hello, > > > > I have two questions about RTPproxy. > > > > Firstly, I want to use Send RTPproxy for communicating 2 > > phones behind NAT. In order to realize this RTPproxy should wait for > > both sides to send their RTPs to RTPproxy. But it starts forwarding > > these RTPs to the other side immediately after receiving from any side. > > Is there any way to do this without giving a fake IP address in Update > > message? ( I want to have a control on IPs by forcing ) > > I am not sure why this scenario is a problem. The proxy will update with > correct IP address once it gets the first packet from the other party. > Can you please elaborate? > > > Secondly, I want to know whether these phones start communicating or > > not. Is it possible for RTPproxy to send notification when peers start > > talking successfully (RTPs are being forwarded from both sides) ? > > There is no such functionality currently, but it should not be difficult > to implement something along the lines. > > Regards, > -- > Maksym Sobolyev > Sippy Software, Inc. > Internet Telephony (VoIP) Experts > T/F: +1-646-651-1110 > Web: http://www.sippysoft.com > MSN: sales at sippysoft.com > Skype: SippySoft > > _______________________________________________ > Users mailing list > Users at rtpproxy.org > http://lists.rtpproxy.org/mailman/listinfo/users _________________________________________________________________ Windows Live: Keep your friends up to date with what you do online. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.rtpproxy.org/pipermail/users/attachments/20100112/05a4c100/attachment.htm From klaus.mailinglists at pernau.at Tue Jan 12 03:47:36 2010 From: klaus.mailinglists at pernau.at (Klaus Darilion) Date: Tue, 12 Jan 2010 12:47:36 +0100 Subject: [RTPproxy Users] notification on both sides start sending their RTPs? In-Reply-To: References: , <4B4C4788.2070407@sippysoft.com> Message-ID: <4B4C6158.4060608@pernau.at> Arman Gungor schrieb: > Imagine the following scenerio, A and B are phones behind NAT. > > A---------------------NAT-------------------------RTPproxy-----------------------NAT-------------------------------B > > RTP----------------> > -------------------> ------------------> BLOCKED > > If A sends RTP packets and RTPproxy forwards them before waiting for B's > RTPs to come, B's NAT device will block that RTP port, then they would > not be able talk at all. > > Also I want to have a control on IPs of A and B for security reasons. > Another person( C ) should not be able send packets to the port reserved > for B. If C can be able to send packets to this reserved port then he > will be able to hear A's voice. Am I right? Depends on the used flags when calling force_rtpproxy. Per default nathelper module will instruct rtpproxy to accept RTP packets only from the public IP of the NAT. You can change behavior with r flag. > > Thanks, > > Arman Gungor > > > > Date: Tue! , 12 Jan 2010 01:57:28 -0800 > > From: sobomax at sippysoft.com > > To: users at rtpproxy.org > > Subject: Re: [RTPproxy Users] notification on both sides start > sending their RTPs? > > > > Arman Gungor wrote: > > > Hello, > > > > > > I have two questions about RTPproxy. > > > > > > Firstly, I want to use Send RTPproxy for > communicating 2 > > > phones behind NAT. In order to realize this RTPproxy should wait for > > > both sides to send their RTPs to RTPproxy. But it starts forwarding > > > these RTPs to the other side immediately after receiving from any > side. > > > Is there any way to do this without giving a fake IP address in Update > > > message? ( I want to have a control on IPs by forcing ) > > > > I am not sure why this scenario is a problem. The proxy will update with > > correct IP address once it gets the first packet from the other party! . > > Can you please elaborate? > > > > > Sec! ondly, I want to know whether these phones start communicating or > > > not. Is it possible for RTPproxy to send notification when peers start > > > talking successfully (RTPs are being forwarded from both sides) ? > > > > There is no such functionality currently, but it should not be difficult > > to implement something along the lines. > > > > Regards, > > -- > > Maksym Sobolyev > > Sippy Software, Inc. > > Internet Telephony (VoIP) Experts > > T/F: +1-646-651-1110 > > Web: http://www.sippysoft.com > > MSN: sales at sippysoft.com > > Skype: SippySoft > > > > _______________________________________________ > > Users mailing list > > Users at rtpproxy.org > > http://lists.rtpproxy.org/mailman/listinfo/users > > ------------------------------------------------------------------------ > Windows Live: Keep your friends up to date with what you do online. > PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010> > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users at rtpproxy.org > http://lists.rtpproxy.org/mailman/listinfo/users From vragukumar at signalogic.com Tue Jan 12 13:09:21 2010 From: vragukumar at signalogic.com (Vikram Ragukumar) Date: Tue, 12 Jan 2010 15:09:21 -0600 Subject: [RTPproxy Users] Rtpproxy not recording session Message-ID: <4B4CE501.3010801@signalogic.com> Hello, Shown below is a high level sketch of the system setup. ----------------------- | CentOS5.4 | ----------- -------- | ---- --- ---- | | CentOS5.4 | |Internet|--|eth0|---|br0|---|eth1|-----| Asterisk | -------- | ---- --- ---- | ----------- | Kamailio + rtpproxy | ----------------------- br0 has a static Public IPv4 address a.b.c.d, and acts as a bridge between eth0 and eth1. Both eth0 and eth1 do not have IPv4 addresses. The box running Asterisk also has a public IPv4 address a.b.c.e The default config file for Kamailio, Kamailo.cfg has been edited to include Nat and mysql support. I need Kamailio to function as a proxy by forwarding all SIP signaling information to Asterisk. Rtpproxy is started using rtpproxy -l a.b.c.d -s udp:127.0.0.1:7722 -r /var/log -a -S /root/rtpproxy then Kamailio is started to listen on 127.0.0.1:5060 and a.b.c.d:5060 I make a test call using x-lite softphone registered with Asterisk, however rtpproxy does not seem to record the session at all. I have checked the directory to make sure the permissions are alright. Is there something i am missing here ? Thanks and Regards, Vikram. From vragukumar at signalogic.com Wed Jan 13 15:35:52 2010 From: vragukumar at signalogic.com (Vikram Ragukumar) Date: Wed, 13 Jan 2010 17:35:52 -0600 Subject: [RTPproxy Users] Converting rtpproxy recorded .rtp file to .wav file Message-ID: <4B4E58D8.3070907@signalogic.com> Hello, I used Kamailio+rtpproxy to record a session and rtpproxy outputs the following files long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp, long_file_name.o.rtcp http://www.rtpproxy.org/wiki/RTPproxy/FAQ From the Rtpproxy FAQ above, i tried to extract the audio using rtpbreak and sox. rtpbreak -W -r long_file_name.a.rtp rtpbreak -W -r long_file_name.o.rtp The above commands generate rtp.0.0.raw, rtp.1.0.raw. Then when i run sox using sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm -s out.wav i get the following errors : sox: invalid option -- - sox: -c must be given a number Is there a switch/anything else that i am missing ? Thanks in advance, Regards, Vikram. From k at isvyaz.ru Wed Jan 13 21:08:15 2010 From: k at isvyaz.ru (Shpinev Konstantin) Date: Thu, 14 Jan 2010 10:08:15 +0500 Subject: [RTPproxy Users] Converting rtpproxy recorded .rtp file to .wav file In-Reply-To: <4B4E58D8.3070907@signalogic.com> References: <4B4E58D8.3070907@signalogic.com> Message-ID: <1853934192.20100114100815@isvyaz.ru> Hello, Vikram. > Hello, > I used Kamailio+rtpproxy to record a session and rtpproxy outputs the > following files > long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp, > long_file_name.o.rtcp > http://www.rtpproxy.org/wiki/RTPproxy/FAQ > From the Rtpproxy FAQ above, i tried to extract the audio using > rtpbreak and sox. > rtpbreak -W -r long_file_name.a.rtp > rtpbreak -W -r long_file_name.o.rtp > The above commands generate rtp.0.0.raw, rtp.1.0.raw. > Then when i run sox using > sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm > -s out.wav i get the following errors : > sox: invalid option -- - > sox: -c must be given a number > Is there a switch/anything else that i am missing ? Try to use '-M' instead of '--combine merge' > Thanks in advance, > Regards, > Vikram. > _______________________________________________ > Users mailing list > Users at rtpproxy.org > http://lists.rtpproxy.org/mailman/listinfo/users -- Shpinev Konstantin mailto:k at isvyaz.ru From vragukumar at signalogic.com Thu Jan 14 11:25:46 2010 From: vragukumar at signalogic.com (Vikram Ragukumar) Date: Thu, 14 Jan 2010 13:25:46 -0600 Subject: [RTPproxy Users] Converting rtpproxy recorded .rtp file to .wav file In-Reply-To: <4B4E58D8.3070907@signalogic.com> References: <4B4E58D8.3070907@signalogic.com> Message-ID: <4B4F6FBA.2060600@signalogic.com> Hello, An update, I tried using sox to convert the two .raw files into 2 mono channel wave files. The command line i used is below : sox -r 8k -b -c 1 -u rtp.0.0.raw rtp0.wav sox -r 8k -b -c 1 -u rtp.1.0.raw rtp1.wav When i listen to the .wav files, i hear speech but it is buried in a lot of noise. During blank periods (periods of no speech) there is a constant volume high pitched noise. Also during periods of speech, there seems to be bursts of noise in the background. The other engineer i work with and i, think that it is possibly because non-speech data is being interpreted as speech. What switch options should i change while invoking sox from the command line to get rid of the noise? Thanks and Regards, Vikram. Vikram Ragukumar wrote: > Hello, > > I used Kamailio+rtpproxy to record a session and rtpproxy outputs the > following files > > long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp, > long_file_name.o.rtcp > > http://www.rtpproxy.org/wiki/RTPproxy/FAQ > From the Rtpproxy FAQ above, i tried to extract the audio using > rtpbreak and sox. > > rtpbreak -W -r long_file_name.a.rtp > rtpbreak -W -r long_file_name.o.rtp > > The above commands generate rtp.0.0.raw, rtp.1.0.raw. > > Then when i run sox using > sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm > -s out.wav i get the following errors : > > sox: invalid option -- - > sox: -c must be given a number > > Is there a switch/anything else that i am missing ? > > Thanks in advance, > Regards, > Vikram. > > From vragukumar at signalogic.com Thu Jan 14 13:49:58 2010 From: vragukumar at signalogic.com (Vikram Ragukumar) Date: Thu, 14 Jan 2010 15:49:58 -0600 Subject: [RTPproxy Users] Rtpproxy recorded .rtp file to .wav format In-Reply-To: References: Message-ID: <4B4F9186.9000906@signalogic.com> Shpinev, Thanks for your response. -M seems to be an unrecognized switch. I have been using rtpbreak to generate the .raw file. Subsequent to which i use sox to convert the .raw to a .wav file. By importing the the .raw file into Hypersignal software, i found that the .raw file doesnt entirely seem to be composed of speech samples, so there could be a problem at the rtpbreak step. This is the command i used to convert rtpproxy's capture file to .raw format rtpbreak -W -r capturefile.rtp Am i missing something here ? Thanks and Regards, Vikram. > Date: Thu, 14 Jan 2010 10:08:15 +0500 > From: Shpinev Konstantin > Subject: Re: [RTPproxy Users] Converting rtpproxy recorded .rtp file > to .wav file > To: RTPproxy Users > Message-ID: <1853934192.20100114100815 at isvyaz.ru> > Content-Type: text/plain; charset=windows-1251 > > Hello, Vikram. > >> Hello, > >> I used Kamailio+rtpproxy to record a session and rtpproxy outputs the >> following files > >> long_file_name.a.rtp, long_file_name.a.rtcp, long_file_name.o.rtp, >> long_file_name.o.rtcp > >> http://www.rtpproxy.org/wiki/RTPproxy/FAQ >> From the Rtpproxy FAQ above, i tried to extract the audio using >> rtpbreak and sox. > >> rtpbreak -W -r long_file_name.a.rtp >> rtpbreak -W -r long_file_name.o.rtp > >> The above commands generate rtp.0.0.raw, rtp.1.0.raw. > >> Then when i run sox using >> sox --combine merge -r 8k -A rtp.0.0.raw -r 8k -A rtp.1.0.raw -t wavpcm >> -s out.wav i get the following errors : > >> sox: invalid option -- - >> sox: -c must be given a number > >> Is there a switch/anything else that i am missing ? > > Try to use '-M' instead of '--combine merge' > > >> Thanks in advance, >> Regards, >> Vikram. > >> _______________________________________________ >> Users mailing list >> Users at rtpproxy.org >> http://lists.rtpproxy.org/mailman/listinfo/users > > From agoengnug at gmail.com Sat Jan 23 07:16:51 2010 From: agoengnug at gmail.com (agung nugroho) Date: Sat, 23 Jan 2010 15:16:51 +0000 Subject: [RTPproxy Users] about video on rtpproxy Message-ID: <1ab0905c1001230716u1bc63363x6712a50e056915bd@mail.gmail.com> hi all,i am new on this mailing-list, :-D i want to share my problem,*just kidding.. i want to ask you all about video on rtpproxy. Here my problem: I build a server with nathelper enable,and rtpproxy on the same machine. When I activated my server, I successful connect to rtpproxy,but went i make a video/audio call between 2 users, I got unconnected with rtpproxy. This is my simple configuration : client IPv6 --> server SIP <-- client IPv4 can anyone help me??? is it need to add another setting on my system or just rtpproxy doesn't support video calling yet.. thanks b4.. *sorry for my bad english. :-D -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.rtpproxy.org/pipermail/users/attachments/20100123/9015398f/attachment.htm From klaus.mailinglists at pernau.at Mon Jan 25 01:01:13 2010 From: klaus.mailinglists at pernau.at (Klaus Darilion) Date: Mon, 25 Jan 2010 10:01:13 +0100 Subject: [RTPproxy Users] about video on rtpproxy In-Reply-To: <1ab0905c1001230716u1bc63363x6712a50e056915bd@mail.gmail.com> References: <1ab0905c1001230716u1bc63363x6712a50e056915bd@mail.gmail.com> Message-ID: <4B5D5DD9.4050108@pernau.at> Does it work using ipv4<-->ipv4 ? klaus agung nugroho schrieb: > hi all,i am new on this mailing-list, :-D > i want to share my problem,*just kidding.. > i want to ask you all about video on rtpproxy. Here my problem: > > I build a server with nathelper enable,and rtpproxy on the same machine. > When I activated my server, I successful connect to rtpproxy,but went i > make a video/audio call between 2 users, I got unconnected with > rtpproxy. This is my simple configuration : > > client IPv6 --> server SIP <-- client IPv4 > > can anyone help me??? is it need to add another setting on my system or > just rtpproxy doesn't support video calling yet.. > > thanks b4.. > > *sorry for my bad english. :-D > > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users at rtpproxy.org > http://lists.rtpproxy.org/mailman/listinfo/users From agoengnug at gmail.com Mon Jan 25 15:17:55 2010 From: agoengnug at gmail.com (agung nugroho) Date: Mon, 25 Jan 2010 23:17:55 +0000 Subject: [RTPproxy Users] Users Digest, Vol 22, Issue 6 In-Reply-To: References: Message-ID: <1ab0905c1001251517h17c0c077g3b93236af1caceac@mail.gmail.com> For IPv4 -- IPv4 client I use another configuration that doesn't load nathelper and not using rtpproxy. Cause if I use my configuration for IPv4 -- IPv4, I get audio and video calling even chatting is available. Did must I try same configuration for transition between IPv4 -- IPv6 in IPv4 -- IPv4 ??? On Mon, Jan 25, 2010 at 8:00 PM, wrote: > Send Users mailing list submissions to > users at rtpproxy.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.rtpproxy.org/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request at rtpproxy.org > > You can reach the person managing the list at > users-owner at rtpproxy.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: about video on rtpproxy (Klaus Darilion) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 25 Jan 2010 10:01:13 +0100 > From: Klaus Darilion > Subject: Re: [RTPproxy Users] about video on rtpproxy > To: RTPproxy Users > Message-ID: <4B5D5DD9.4050108 at pernau.at> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Does it work using ipv4<-->ipv4 ? > > klaus > > agung nugroho schrieb: > > hi all,i am new on this mailing-list, :-D > > i want to share my problem,*just kidding.. > > i want to ask you all about video on rtpproxy. Here my problem: > > > > I build a server with nathelper enable,and rtpproxy on the same machine. > > When I activated my server, I successful connect to rtpproxy,but went i > > make a video/audio call between 2 users, I got unconnected with > > rtpproxy. This is my simple configuration : > > > > client IPv6 --> server SIP <-- client IPv4 > > > > can anyone help me??? is it need to add another setting on my system or > > just rtpproxy doesn't support video calling yet.. > > > > thanks b4.. > > > > *sorry for my bad english. :-D > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users at rtpproxy.org > > http://lists.rtpproxy.org/mailman/listinfo/users > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users at rtpproxy.org > http://lists.rtpproxy.org/mailman/listinfo/users > > > End of Users Digest, Vol 22, Issue 6 > ************************************ > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.rtpproxy.org/pipermail/users/attachments/20100125/d48b940b/attachment.htm From agoengnug at gmail.com Mon Jan 25 15:21:31 2010 From: agoengnug at gmail.com (agung nugroho) Date: Mon, 25 Jan 2010 23:21:31 +0000 Subject: [RTPproxy Users] about video on rtpproxy Message-ID: <1ab0905c1001251521n1ef44d5sd59c21b10789de12@mail.gmail.com> For IPv4 -- IPv4 client I use another configuration that doesn't load nathelper and not using rtpproxy. Cause if I use my configuration for IPv4 -- IPv4, I get audio and video calling even chatting is available. Did must I try same configuration for transition between IPv4 -- IPv6 in IPv4 -- IPv4 ??? *sorry for double post. On Mon, Jan 25, 2010 at 8:00 PM, wrote: > Send Users mailing list submissions to > users at rtpproxy.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.rtpproxy.org/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request at rtpproxy.org > > You can reach the person managing the list at > users-owner at rtpproxy.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: about video on rtpproxy (Klaus Darilion) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 25 Jan 2010 10:01:13 +0100 > From: Klaus Darilion > Subject: Re: [RTPproxy Users] about video on rtpproxy > To: RTPproxy Users > Message-ID: <4B5D5DD9.4050108 at pernau.at> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Does it work using ipv4<-->ipv4 ? > > klaus > > agung nugroho schrieb: > > hi all,i am new on this mailing-list, :-D > > i want to share my problem,*just kidding.. > > i want to ask you all about video on rtpproxy. Here my problem: > > > > I build a server with nathelper enable,and rtpproxy on the same machine. > > When I activated my server, I successful connect to rtpproxy,but went i > > make a video/audio call between 2 users, I got unconnected with > > rtpproxy. This is my simple configuration : > > > > client IPv6 --> server SIP <-- client IPv4 > > > > can anyone help me??? is it need to add another setting on my system or > > just rtpproxy doesn't support video calling yet.. > > > > thanks b4.. > > > > *sorry for my bad english. :-D > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Users mailing list > > Users at rtpproxy.org > > http://lists.rtpproxy.org/mailman/listinfo/users > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users at rtpproxy.org > http://lists.rtpproxy.org/mailman/listinfo/users > > > End of Users Digest, Vol 22, Issue 6 > ************************************ > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.rtpproxy.org/pipermail/users/attachments/20100125/6f193408/attachment.htm From vgarzon at hotmail.es Mon Jan 25 19:57:15 2010 From: vgarzon at hotmail.es (=?iso-8859-1?B?Vu1jdG9yIEdhcnrzbiBNYXLtbg==?=) Date: Mon, 25 Jan 2010 22:57:15 -0500 Subject: [RTPproxy Users] opensips help Message-ID: hello, i'm working in a project when i need to install music on hold in opensips, i've read that i can do this by using RTPproxy, but i need to know how i can install the RTPproxy support for opensips. Can you help me? thanks _________________________________________________________________ Connect to the next generation of MSN Messenger? http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=wlmailtagline -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.rtpproxy.org/pipermail/users/attachments/20100125/7f9612a8/attachment.htm From vragukumar at signalogic.com Tue Jan 26 11:35:38 2010 From: vragukumar at signalogic.com (Vikram Ragukumar) Date: Tue, 26 Jan 2010 13:35:38 -0600 Subject: [RTPproxy Users] Rtpproxy transcoder patch Message-ID: <4B5F440A.7020704@signalogic.com> Hello, I came across a post by Stefan Sayer regarding an rtpproxy transcoder patch http://lists.iptel.org/pipermail/serdev/2008-July/012794.html. We are interested in having rtpproxy perform transcoding, both in software and using DSP array cards, and are currently running rtpproxy version 1.2.1. Is this link http://user.cs.tu-berlin.de/~sayer/transcoder/ that is mentioned on the post our starting point or is there an updated version ? Thanks and Regards, Vikram. From ibc at aliax.net Wed Jan 27 05:57:23 2010 From: ibc at aliax.net (=?utf-8?q?I=C3=B1aki_Baz_Castillo?=) Date: Wed, 27 Jan 2010 14:57:23 +0100 Subject: [RTPproxy Users] rtpproxy says "open file descriptors limit = 1024" but it should be 16384 Message-ID: <201001271457.23891.ibc@aliax.net> Hi, I run rtpproxy as follos: rtpproxy -l XXX.XXX.XXX.XXX -s unix:/var/run/rtpproxy.sock -m 30000 -M 40000 -u kamailio kamailio -d INFO:LOG_LOCAL6 -t -1 User "kamailio" can open up to 16384 files: $ su kamailio -s /bin/bash -c "ulimit -n" 16384 However under high load I get the following warning in rtpproxy logs: WARN:handle_command: passed 80% threshold on the open file descriptors limit (1024), consider increasing the limit using -L command line option Why does it think that its limit is 1024 rather than the real value 16384? The server has been recently restarted so for sure the value configured in /etc/security/limits.conf is the value all the process have started with (I use Linux Debian). $ rtpproxy -v Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Thanks a lot. -- I?aki Baz Castillo From ibc at aliax.net Wed Jan 27 07:00:30 2010 From: ibc at aliax.net (=?utf-8?q?I=C3=B1aki_Baz_Castillo?=) Date: Wed, 27 Jan 2010 16:00:30 +0100 Subject: [RTPproxy Users] rtpproxy says "open file descriptors limit = 1024" but it should be 16384 In-Reply-To: <201001271457.23891.ibc@aliax.net> References: <201001271457.23891.ibc@aliax.net> Message-ID: <201001271600.30738.ibc@aliax.net> El Mi?rcoles, 27 de Enero de 2010, I?aki Baz Castillo escribi?: > Hi, I run rtpproxy as follos: > > rtpproxy -l XXX.XXX.XXX.XXX -s unix:/var/run/rtpproxy.sock -m 30000 -M > 40000 -u kamailio kamailio -d INFO:LOG_LOCAL6 -t -1 > > > User "kamailio" can open up to 16384 files: > > $ su kamailio -s /bin/bash -c "ulimit -n" > 16384 > > However under high load I get the following warning in rtpproxy logs: > > WARN:handle_command: passed 80% threshold on the open file descriptors > limit (1024), consider increasing the limit using -L command line option Perhaps RtpProxy automatically sets "L" to 1024 if the parameter is missing? -- I?aki Baz Castillo From sobomax at sippysoft.com Wed Jan 27 11:12:24 2010 From: sobomax at sippysoft.com (Maxim Sobolev) Date: Wed, 27 Jan 2010 11:12:24 -0800 Subject: [RTPproxy Users] rtpproxy says "open file descriptors limit = 1024" but it should be 16384 In-Reply-To: <201001271600.30738.ibc@aliax.net> References: <201001271457.23891.ibc@aliax.net> <201001271600.30738.ibc@aliax.net> Message-ID: <4B609018.6040403@sippysoft.com> I?aki Baz Castillo wrote: > El Mi?rcoles, 27 de Enero de 2010, I?aki Baz Castillo escribi?: >> Hi, I run rtpproxy as follos: >> >> rtpproxy -l XXX.XXX.XXX.XXX -s unix:/var/run/rtpproxy.sock -m 30000 -M >> 40000 -u kamailio kamailio -d INFO:LOG_LOCAL6 -t -1 >> >> >> User "kamailio" can open up to 16384 files: >> >> $ su kamailio -s /bin/bash -c "ulimit -n" >> 16384 >> >> However under high load I get the following warning in rtpproxy logs: >> >> WARN:handle_command: passed 80% threshold on the open file descriptors >> limit (1024), consider increasing the limit using -L command line option > > Perhaps RtpProxy automatically sets "L" to 1024 if the parameter is missing? Hmm, no it doesn't. More likely is that setuid(2) system call is not setting limits, I will check that and get back to you. Regards, -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts T/F: +1-646-651-1110 Web: http://www.sippysoft.com MSN: sales at sippysoft.com Skype: SippySoft