July 2012 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] FreeSWITCH Conference Call Tomorrow
by Michael Collins 10 Jul '12

10 Jul '12

Hello Community! Just a friendly reminder: our conference call is scheduled for 1PM Eastern/10AM Pacific time tomorrow. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_11 Dave Kompel (drk__) will be discussing Windows firewall security. Rumor has it that he's got an interesting example of how to do server push notification for those of you who want to see an example of how to do server updates to a browser. On the ClueCon front we just wanted to remind those of you who will be in Chicago for ClueCon that Anthony will be enjoying birthday number 40-something, so plan accordingly. :) Thanks! -MC

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[Freeswitch-dev] It's That Time Again: Buy Dinner For The FreeSWITCH Developers
by Michael Collins 10 Jul '12

10 Jul '12

Greetings all! It has been quite some time since we all got together and pitched in to buy dinner for the FreeSWITCH development team. A well-fed dev team means more features and fewer bugs, so let's all head over to the main FreeSWITCH page<http://www.freeswitch.org?feed12>and hit the donate button. Be sure to click the "special instructions" link in Paypal and mention that your donation is for feeding the developers. Feel free to be creative. :) If you prefer to use a payment method other than FreeSWITCH then please contact Brian West <brian(a)freeswitch.org>directly. Thanks again for being such a great community and support us over the years!

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[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 09 Jul '12

09 Jul '12

Greetings all! We are back to a full week after taking last Wednesday off. This week<http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_11?cc12-0709>we have Dave Kompel scheduled to talk to us on a subject that has been of much interest in recent weeks: firewall security on Windows deployments of FreeSWITCH. Many of you know that we have fail2ban<http://wiki.freeswitch.org/wiki/Fail2ban?cc12-0709>available for Linux/Unix based installs, but what about Windows <http://xkcd.com/528?cc12-0709>? There are indeed techniques for Windows systems to update the Windows firewall to prevent things like the friendly scanner <http://www.freeswitch.org/node/343?cc12-0709>. Recall: the friendly scanner is an abuse of a SIP testing tool called SIPVicious<http://blog.sipvicious.org?cc12-0709>. People use it for nefarious purposes, so having the ability to lock down the source IP address on the fly is a powerful technique to employ. We look forward to this discussion <http://xkcd.com/841/>. We are happy to report that we have a new GOLD sponsor at ClueCon: Flowroute<http://www.flowroute.com?cc12-0709>! The folks at Flowroute have graciously offered to make sure that our attendees are well taken care of. First, they are supplying the gift bags that all of our attendees will receive. In those bags will be lots of goodies from Flowroute and our other sponsors<http://www.cluecon.com/sponsors?cc12-0709>. A special item will be a $25 gift card for Flowroute. Additionally, Flowroute is sponsoring a pizza party on Monday night, August 6 at Gino's<http://www.ginoseast.com?cc12-0709>in downtown Chicago! We hope you all will be able to join us. Only a few more weeks until we're all together again! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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[Freeswitch-dev] How to build Debian packages with Sangoma module included?
by Craig Ayliffe 04 Jul '12

04 Jul '12

Hi, Just wondering if anyone has been able to build Debian packages of FS with mod_sangoma_codec package built as well? I have removed it from the list of avoid_mods in debian/boot, and it goes to build it but fails because it can't find the Sangoma libraries when compiling. I am not sure how to get those libraries included in the fakeroot when it builds the packages. Any hints/assistance would be great thanks (Sorry if this should be in the user list but assumed this was the right one) -- Craig Ayliffe

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[Freeswitch-dev] rtp question
by Seven Du 04 Jul '12

04 Jul '12

Hi, libx264 encoded NALUs exceeding packet window and some phones don't like it so I need to manually split with FU-A, I tried frame->m but looks like all packets has marker = 1 even frame->m = 0; by looking at mod_fsv.c, I added the following and generate my own rtp header switch_set_flag((&vid_frame), SFF_RAW_RTP); switch_set_flag((&vid_frame), SFF_PROXY_PACKET); but looks like it doens't send my rtp header, switch_rtp.c has a check also on SWITCH_RTP_FLAG_RAW_WRITE which I don't see in mod_fsv.c fwd = (switch_test_flag(rtp_session, SWITCH_RTP_FLAG_RAW_WRITE) && switch_test_flag(frame, SFF_RAW_RTP)) ? 1 : 0; I also tried set SFF_RTP_HEADER, but it tends to send double as packets looks if (switch_test_flag(frame, SFF_RTP_HEADER)) { in switch_rtp.c doesn't stop and what the purpose of this param? Any highlights? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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Re: [Freeswitch-dev] Question about RTP marker
by Peter Olsson 04 Jul '12

04 Jul '12

I've added FS-4382, please check there for further info. /Peter -----Ursprungligt meddelande----- Från: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För Peter Olsson Skickat: den 3 juli 2012 18:07 Till: freeswitch-dev(a)lists.freeswitch.org Ämne: Re: [Freeswitch-dev] Question about RTP marker Thank you for your replies, both Tony and Kristian - it made things a little bit clearer for me :) I will submit a proposed patch to Jira later on (probably tomorrow), and you can review my changes there, it's a small change. /Peter -----Ursprungligt meddelande----- Från: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För Kristian Kielhofner Skickat: den 3 juli 2012 17:33 Till: freeswitch-dev(a)lists.freeswitch.org Ämne: Re: [Freeswitch-dev] Question about RTP marker +1 The use of the RTP marker has been broken and misunderstood by many implementations... The references the OP found about "talkspurts" are probably referring to the use of markers with VAD (voice activity detection). When VAD detects speech the first RTP packet is supposed to have the marker set, almost as a way of saying "Hey! I'm talking again". Of course with VAD a new talkspurt will also have a different timestamp... FreeSWITCH (out of respect for jitter buffers, as Anthony said) tries to pass original (read: meaningful) timestamps and use markers wherever possible. Consider the following scenario: Call -> IVR (FS starts generating RTP) IVR -> bridge to external SIP device (device starts generating RTP) In this scenario FS will generate the initial RTP stream with it's own timestamps. When the call is bridged to the remote device FS will set the marker and pass the timestamps from the B leg to the A leg. This allows the far end jitterbuffers (not FS) to buffer jitter based on the actual end to end RTP stream. Unfortunately this can cause all kinds of confusion with broken implementations. While changing the SSRC is (of course) a hack if it improves compatibility while maintaining the usefulness and integrity of timestamps (and jitter buffers) I'm for it. On Tue, Jul 3, 2012 at 11:18 AM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: > The line is really blurry between what is right and what works better. > I think the only real rule about the marker bit is that you are > supposed to set it any time the timestamp is not the next expected > timestamp. > > This is supposed to help you to understand that you are purposely > sending a packet that has a different timestamp base or that you are > resuming after intentionally not sending packets for a period of time. > This tells the remote end to flush its jitter buffer which probably > explains the drop in audio. > > If changing the SSRC is an acceptable workaround for when the rtp > streams change that do not result in audio gaps, I'm more than happy > to accept it as a default and we can always make it a config option if > it presents any problems. > > > > On Tue, Jul 3, 2012 at 9:39 AM, Peter Olsson > <peter.olsson(a)visionutveckling.se> wrote: >> When I've been debugging RTP streams in Wireshark, I've noticed that >> Whireshark will get "everything wrong" when the timestamp changes (is >> reset). For instance, if I call a conference, by the time I get into >> the conference, the RTP timestamps will start over, and FS will send >> the marker bit By that time Wireshark will report strange figures for jitter etc. >> >> >> >> After some Googling I've found that the meaning of the RTP marker is >> to indicate "the beginning of a talkspurt". Which is not very clear >> to me... :) >> >> >> >> So my question is - is RTP marker really enough when starting a "new >> RTP stream" (not really new, but since the timestamps are reset we >> can think of it like that) - according to what I've been able to >> find, maybe the SSRC should also be changed? >> >> >> >> I have created a patch that will change the SSRC when a reset like >> this occurs, and after that Wireshark detects my audio streams >> without problems (though, it will detect more streams), but no strange jitter figures etc. >> are reported. >> >> >> >> The reason I started digging into this was because a PBX I was >> connecting to lost about the first second of audio after connecting >> to the conference, after also changing the SSRC the problem seems resolved. >> >> >> >> I could make a RTP_BUG setting for this, but I'm not really sure if I >> should really care - that's why I'm asking here, if anyone has more >> knowledge about this? >> >> >> >> /Peter >> >> >> >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de >> v >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-dev mailing list FreeSWITCH-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-dev mailing list FreeSWITCH-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org !DSPAM:4ff3184a32761575917611!

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[Freeswitch-dev] ClueCon 2012 - Last Chance for 16 Entries
by Michael Collins 03 Jul '12

03 Jul '12

Hello all! I am happy to report that a number of people have taken advantage of this opportunity and have registered in time to get their 16 entries. Remember that you can register <http://www.cluecon.com/register?cc12-0703>any time on July 3rd and still qualify for the extra entries. The bits will shift at midnight Central time so don't delay! Only 34 more days and we'll all be in Chicago. See you then! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Re: [Freeswitch-dev] Question about RTP marker
by Peter Olsson 03 Jul '12

03 Jul '12

Thank you for your replies, both Tony and Kristian - it made things a little bit clearer for me :) I will submit a proposed patch to Jira later on (probably tomorrow), and you can review my changes there, it's a small change. /Peter -----Ursprungligt meddelande----- Från: freeswitch-dev-bounces(a)lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För Kristian Kielhofner Skickat: den 3 juli 2012 17:33 Till: freeswitch-dev(a)lists.freeswitch.org Ämne: Re: [Freeswitch-dev] Question about RTP marker +1 The use of the RTP marker has been broken and misunderstood by many implementations... The references the OP found about "talkspurts" are probably referring to the use of markers with VAD (voice activity detection). When VAD detects speech the first RTP packet is supposed to have the marker set, almost as a way of saying "Hey! I'm talking again". Of course with VAD a new talkspurt will also have a different timestamp... FreeSWITCH (out of respect for jitter buffers, as Anthony said) tries to pass original (read: meaningful) timestamps and use markers wherever possible. Consider the following scenario: Call -> IVR (FS starts generating RTP) IVR -> bridge to external SIP device (device starts generating RTP) In this scenario FS will generate the initial RTP stream with it's own timestamps. When the call is bridged to the remote device FS will set the marker and pass the timestamps from the B leg to the A leg. This allows the far end jitterbuffers (not FS) to buffer jitter based on the actual end to end RTP stream. Unfortunately this can cause all kinds of confusion with broken implementations. While changing the SSRC is (of course) a hack if it improves compatibility while maintaining the usefulness and integrity of timestamps (and jitter buffers) I'm for it. On Tue, Jul 3, 2012 at 11:18 AM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: > The line is really blurry between what is right and what works better. > I think the only real rule about the marker bit is that you are > supposed to set it any time the timestamp is not the next expected > timestamp. > > This is supposed to help you to understand that you are purposely > sending a packet that has a different timestamp base or that you are > resuming after intentionally not sending packets for a period of time. > This tells the remote end to flush its jitter buffer which probably > explains the drop in audio. > > If changing the SSRC is an acceptable workaround for when the rtp > streams change that do not result in audio gaps, I'm more than happy > to accept it as a default and we can always make it a config option if > it presents any problems. > > > > On Tue, Jul 3, 2012 at 9:39 AM, Peter Olsson > <peter.olsson(a)visionutveckling.se> wrote: >> When I've been debugging RTP streams in Wireshark, I've noticed that >> Whireshark will get "everything wrong" when the timestamp changes (is >> reset). For instance, if I call a conference, by the time I get into >> the conference, the RTP timestamps will start over, and FS will send >> the marker bit By that time Wireshark will report strange figures for jitter etc. >> >> >> >> After some Googling I've found that the meaning of the RTP marker is >> to indicate "the beginning of a talkspurt". Which is not very clear >> to me... :) >> >> >> >> So my question is - is RTP marker really enough when starting a "new >> RTP stream" (not really new, but since the timestamps are reset we >> can think of it like that) - according to what I've been able to >> find, maybe the SSRC should also be changed? >> >> >> >> I have created a patch that will change the SSRC when a reset like >> this occurs, and after that Wireshark detects my audio streams >> without problems (though, it will detect more streams), but no strange jitter figures etc. >> are reported. >> >> >> >> The reason I started digging into this was because a PBX I was >> connecting to lost about the first second of audio after connecting >> to the conference, after also changing the SSRC the problem seems resolved. >> >> >> >> I could make a RTP_BUG setting for this, but I'm not really sure if I >> should really care - that's why I'm asking here, if anyone has more >> knowledge about this? >> >> >> >> /Peter >> >> >> >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-de >> v >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > googletalk:conf+888@conference.freeswitch.org > pstn:+19193869900 > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-dev mailing list FreeSWITCH-dev(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org !DSPAM:4ff3106232761496470876!

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[Freeswitch-dev] Question about RTP marker
by Peter Olsson 03 Jul '12

03 Jul '12

When I've been debugging RTP streams in Wireshark, I've noticed that Whireshark will get "everything wrong" when the timestamp changes (is reset). For instance, if I call a conference, by the time I get into the conference, the RTP timestamps will start over, and FS will send the marker bit By that time Wireshark will report strange figures for jitter etc. After some Googling I've found that the meaning of the RTP marker is to indicate "the beginning of a talkspurt". Which is not very clear to me... :) So my question is - is RTP marker really enough when starting a "new RTP stream" (not really new, but since the timestamps are reset we can think of it like that) - according to what I've been able to find, maybe the SSRC should also be changed? I have created a patch that will change the SSRC when a reset like this occurs, and after that Wireshark detects my audio streams without problems (though, it will detect more streams), but no strange jitter figures etc. are reported. The reason I started digging into this was because a PBX I was connecting to lost about the first second of audio after connecting to the conference, after also changing the SSRC the problem seems resolved. I could make a RTP_BUG setting for this, but I'm not really sure if I should really care - that's why I'm asking here, if anyone has more knowledge about this? /Peter

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[Freeswitch-dev] Reminder: Last chance for 16 tickets before the bits shift!
by Michael Collins 02 Jul '12

02 Jul '12

Just a friendly reminder: July 3rd is the last day to get all 16 tickets in the great ClueCon giveway. Be sure to get registered<http://www.cluecon.com/register?cc12-0702a>by the end of the day on Tuesday, July 3rd to maximize your chance to win big! Looking forward to seeing you all in Chicago next month. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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