- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Status of VP8 codec in Freeswitch
by Muhammad Shahzad 23 Jul '12

23 Jul '12

A patch for VP8 codec pass-through support was submitted in April 09, 2012 by Seven Du, but it is still not committed to FS trunk. Can anyone update us on its status? http://jira.freeswitch.org/browse/FS-4092 If the patch works then i strongly recommend it should be committed to FS trunk, so we can have better integration with WebRTC clients. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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FreeSWITCH Weekly News and Notes
by Michael Collins 23 Jul '12

23 Jul '12

Hello all! Last week we enjoyed a nice conference call with Darren Schreiber<http://www.cluecon.com/speaker/98?cc12-0723>from 2600hz <http://www.2600hz.com?cc12-0723>. Darren gave us a follow up<http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_18?cc12-0723>presentation to the original SIP 101 discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_20?cc12-0723>that was held in June. This was a great presentation that went beyond the basics of SIP. We discussed important concepts like Via and Record Route headers and NOTIFY messages. SIP can do a lot, and with these two presentations we really have only scratched the surface. If we all bug Darren enough then perhaps we'll get a third presentation! The audio of the presentation is in the usual place<http://torrents.freeswitch.org/conf_call_2012-07-18.torrent?cc12-0723>and the slides will be available shortly. This week we will be hearing from the community. We will have a brief report from Dave Kompel who will get us up to speed on using Windows Advanced Firewall to implement Fail2Ban<http://wiki.freeswitch.org/wiki/Fail2ban?cc12-0723>-like functionality. After Dave speaks we will be opening things up for community discussion. Please bring your questions and topics! ClueCon 2012 <http://www.cluecon.com?cc12-0723> is shaping up nicely! We have recently added Ditech Networks<http://www.ditechnetworks.com?cc12-0723> , Bandwidth.com <http://www.bandwidth.com?cc12-0723>, and Yealink<http://www.yealink.com?cc12-0723>as sponsors. We are also happy to announce that Darrell Hensley <http://www.youtube.com/watch?v=sLBLCcSiAr0> will be speaking on the subject of Transitioning to Professional Voice<http://www.cluecon.com/presentation/transition_pro_voice?cc12-0723>. Darrell is the CEO of GMVoices <http://www.gmvoices.com?cc12-0723>, a ClueCon media sponsor for many years and the supplier of Callie<http://www.youtube.com/watch?v=IfkRNgM1IiA>, the voice of FreeSWITCH. We look forward to seeing Darrell in person again. As a quick reminder, if you register<http://www.cluecon.com/register?cc12-0723>by the end of the day on July 25th you will still receive four chances to win in the great ClueCon giveaway. See you in two weeks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0723

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What is the purpose of BLOCK in switch_event.c
by zhangkai.fswitch 23 Jul '12

23 Jul '12

hi, Recently I read the codes in switch_event.c, and I am confused about the purpose of BLOCK. There are 2 locks defined in switch_event.c: static switch_mutex_t *BLOCK = NULL; static switch_thread_rwlock_t *RWLOCK = NULL; My understanding is the RWLOCK is wrlocked when binding or unbinding an event callback function, and rdlocked when executing the callback. So the RWLOCK is enough to protect multi-threads from modifying the event binding list at the same time, why we need another mutex BLOCK? And in the codes, when never the mutex BLOCK is requested to be locked, the RWLOCK must already be locked for write. like this: switch_thread_rwlock_wrlock(RWLOCK); switch_mutex_lock(BLOCK); So if BLOCK is mean to protect something, the RWLOCK is already guaranteed the exclusiveness. Please tell me if I am wrong. Regards, Kai ZHANG

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mod_dingaling does not set crypto
by Richard Screene 21 Jul '12

21 Jul '12

Hello, Using mod_dingaling I am unable to get it to append the crypto details to the session-accept Jingle stanza. The message is sent as: <iq xmlns="jabber:client" from="freeswitch(a)macbooks-macbook.local/talk" to="fff(a)macbooks-macbook.local/talk" type="set" id="300"> <jin:jingle xmlns:jin="urn:xmpp:jingle:1" action="session-accept" sid="233007325" initiator="fff(a)macbooks-macbook.local"> <jin:content name="audio" creator="initiator"> <rtp:description xmlns:rtp="urn:xmpp:jingle:apps:rtp:1" media="audio" ssrc="1351525407"> <rtp:payload-type id="0" name="PCMU" clockrate="8000" ptime="8000" maxptime="8000"> <rtp:parameter name="bitrate" value="64000"></rtp:parameter> </rtp:payload-type> </rtp:description> <p:transport xmlns:p="http://www.google.com/transport/p2p"></p:transport> </jin:content> </jin:jingle> <ses:session xmlns:ses="http://www.google.com/session" type="accept" id="233007325" initiator="fff(a)macbooks-macbook.local"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type> </pho:description> </ses:session> </iq> The full Wireshark trace can be found at http://filebin.ca/8JFB0hyFdHf On investigation it would seem that the TFLAG_SECURE flag is not being set. If I hack out the test on this flag in mdl_build_crypto() then the crypto is included in the message generated by Freeswitch: <iq xmlns="jabber:client" from="freeswitch(a)macbooks-macbook.local/talk" to="fff(a)macbooks-macbook.local/talk" type="set" id="301"> <jin:jingle xmlns:jin="urn:xmpp:jingle:1" action="session-accept" sid="7FnyqtQPPMNPn5h7nnJuRiv3GWg/bRAs" initiator="audio"> <jin:content name="audio" creator="initiator"> <rtp:description xmlns:rtp="urn:xmpp:jingle:apps:rtp:1" media="audio" ssrc="1351524387"> <rtp:payload-type id="0" name="PCMU" clockrate="8000" ptime="8000" maxptime="8000"> <rtp:parameter name="bitrate" value="64000"></rtp:parameter> </rtp:payload-type> <rtp:encryption> <rtp:crypto crypto-suite="AES_CM_128_HMAC_SHA1_80" key-params="inline:J8/3QjLtugHFof08KBIevXxUOBo3x8PdphPMx7J4" tag="1"></rtp:crypto> </rtp:encryption> </rtp:description> <p:transport xmlns:p="http://www.google.com/transport/p2p"></p:transport> </jin:content> </jin:jingle> <ses:session xmlns:ses="http://www.google.com/session" type="accept" id="7FnyqtQPPMNPn5h7nnJuRiv3GWg/bRAs" initiator="audio"> <pho:description xmlns:pho="http://www.google.com/session/phone"> <pho:payload-type id="0" name="PCMU" clockrate="8000" bitrate="64000" xmlns:pho="http://www.google.com/session/phone"></pho:payload-type> </pho:description> </ses:session> </iq> But, I can't see how this flag is ever set. Is this a bug or am I missing something very simple?? This is using Freeswitch commit fa1276d9492fd39ea733f7af301faed3556c9544 (dated 8th July 2012) The jingle profile is here http://filebin.ca/8JGG8ZhEUSh. Regards, Richard

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Update On Donations For H.323
by Michael Collins 21 Jul '12

21 Jul '12

We have received several donations already - thank you very much! Just a reminder: please be sure to leave a note with your donation so that we know what your donation is for. Our target for getting Robert Jongbloed to Chicago from Australia is US $3000. We will let you know when we've met our goal and we'll give you status updates along the way. Thanks again for your generosity! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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How You Can Help With H.323 In FreeSWITCH
by Michael Collins 19 Jul '12

19 Jul '12

Greetings! We would like to let you know about an opportunity we all have to assist with H.323 support in FreeSWITCH. We are working on getting Robert Jongbloed, author of mod_opal and all around cool guy, to ClueCon 2012. Robert does a lot of work with H.323 and his presence with us at ClueCon would go a long way in helping us get H.323 fully implemented in FreeSWITCH. The challenge is that Robert lives in Australia! To that end we are taking up a collection to help defray the substantial travel costs. Here's what you can do: visit FreeSWITCH.org and click the Donate button. In your Paypal donation, click "Add special instructions to the seller" and make a note that this donation is to assist with getting Robert to ClueCon. We'll take care of the rest. Thanks again for all of your wonderful support! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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FreeSWITCH Community Conference Call Reminder
by Michael Collins 18 Jul '12

18 Jul '12

Hello all, Just a reminder that the conference call will start in less than an hour. Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_18 Talk to you soon, Michael

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Need French-Candian translator
by Michael Collins 18 Jul '12

18 Jul '12

Bonjour! We could use some assistance with a few sounds that need to be translated from English to French. These will be prompts that are recorded and made available to the community at large. Please email me off list if you are able to assist with translating about 20 or so phrases. Thanks, Michael

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Multi-server registration with replication
by Tom Parrott 16 Jul '12

16 Jul '12

Hi All, I am have been running Freeswitch for a couple of years in production as a pure voice switch, without any registrations. I have now been tasked with setting up a phone system for our business across 3 offices (in 3 countries). We have SIP trunks with telecoms providers in each country (UK, US and ZA), and want to ensure that if a user dials a local number in each country that the calls are not routed to a central server first, but go out within the same country. I would also like internal extensions to work across the 3 countries, and ideally presence as well. We are using TCP SIP registrations as we have found these more reliable when users are connecting through home routers (or MS ISA firewall) which tend to mangle UDP SIP packets. Whilst I have read on the Freeswitch wiki that you can convert the presence and registration databases to MySQL and replicate them, I am unsure whether this would work when using TCP SIP, as surely the active TCP connection is with the local registration server, and so a registration server in another country could not route calls to the client. This lead me to thinking, perhaps I could use mod_event_zeromq to gather local registration events and forward them to agents on the other servers that would re-write the dialstring and insert a modified registration entry into the local server databases. This way it would cause Freeswitch to route calls for the user that occur on a remote server to the local server and use the existing TCP connection to deliver the call? I have been told I would be able open source this project. If anyone can give any pointers it would be much appreciated. Thanks Tom

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FreeSWITCH Weekly News and Notes
by Michael Collins 16 Jul '12

16 Jul '12

One week closer to ClueCon! Things have been very busy for the FreeSWITCH team and community. In addition to gearing up for ClueCon <http://www.cluecon.com?cc12-0716> we've all been working on our day jobs and mixing in the occasional summer vacation. We've also had inclement weather play a role in scheduling one of our weekly conference<http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0716>call presentations. Dave Kompel, one of our resident Windows gurus, had been scheduled to discuss a technique for handling the Windows firewall in a manner similar to how we use Fail2Ban<http://wiki.freeswitch.org/wiki/Fail2ban?cc12-0716>in a Linux environment, however thunderstorms in his area required his full attention on protecting and keeping his servers up and running. We will reschedule as soon as possible. This week we have scheduled Darren Schreiber from 2600hz<http://www.2600hz.com?cc12-0716>. He will be doing a follow up to his SIP 101 discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_20?cc12-0716>from several weeks ago. We will be diving a bit deeper into the SIP protocol. In the initial discussion we focused most of our attention on getting SIP endpoints registered and the setup/teardown process. This week we will look at some of the other things SIP can do, like presence. We invite you to join us this Wednesday at 1PM Eastern, 10AM Pacific. In ClueCon news we are very happy to report that 2600hz is back as a Silver sponsor! In fact, the aforementioned Darren Schreiber will be giving a presentation<http://www.cluecon.com/presentation/open-source-revamping-telecom-you-want-…>at ClueCon, discussing some of the hard-earned lessons that 2600hz has learned over the past few years. If you are interested in cloud telephony then definitely be there for Darren's talk. For those who've not registered yet please keep in mind that the "bits will shift again" this Wednesday and registrants will receive four entries instead of the current eight. Register now <http://www.cluecon.com/register?cc12-0716> to maximize your opportunity to win lots of great prizes. The ClueCon countdown is already down to 21 days! We are looking forward to seeing everyone together again in Chicago. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0716

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