- FreeSWITCH-users - FreeSWITCH Mailing Lists

[CFP] FOSDEM 2016, RTC devroom, speakers, volunteers neeeded
by Daniel Pocock 30 Oct '15

30 Oct '15

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2016 takes place 30-31 January 2016 in Brussels, Belgium. http://fosdem.org This email contains information about: - Real-Time communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (including the Saturday night dinner), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Saturday, 30 January 2016. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc Speaking opportunities ---------------------- Note: if you used Pentabarf before, please use the same account/username Main track: the deadline for main track presentations is midnight today, 30 October. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit Real-Time Communications dev-room: deadline 27 November Please also use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real-Time devroom". https://penta.fosdem.org/submission/FOSDEM16/ Other dev-rooms: some speakers may find their topic is in the scope of more than one dev-room. It is permitted to apply to more than one dev-room but please be kind enough to tell us if you do this. You can find the full list of dev-rooms at https://www.fosdem.org/2016/schedule/tracks/ Lightning talks: deadline 27 November The lightning talks are an excellent opportunity to introduce a wider audience to your project. Given that dev-rooms are becoming increasingly busy, all speakers are encouraged to consider applying for a lightning talk as well as a slot in the dev-room. On the "General" tab, please look for the "Track" option and choose "Lightning Talks". https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. For any questions, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 30 January - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists FOSDEM is made possible by volunteers and if you have time to contribute, please feel free to get involved through https://volunteers.fosdem.org/ Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on 28 and 29 January 2016 http://wiki.xmpp.org/web/Summit_19 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit We are also considering a more general RTC or telephony summit, potentially on 29 January. Please join the Free-RTC mailing list and send an email if you would be interested in participating, sponsoring or hosting such an event. Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 29 January On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: http://planet.jabber.org ralphm(a)ik.nu http://planet.sip5060.net daniel(a)pocock.pro http://planet.opentelecoms.org daniel(a)pocock.pro Please also link to the Planet sites from your own blog or web site. Contact ======= For discussion and queries, please join the free-rtc mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Daniel Pocock <daniel(a)pocock.pro> Ralph Meijer <ralphm(a)ik.nu> Saúl Ibarra Corretgé <saul(a)ag-projects.com> Iain R. Learmonth <irl(a)debian.org>

1 0

ClueCon Weekly – October 28, 2015 – Brian West
by Ken Rice 30 Oct '15

30 Oct '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1LGF0FZ ClueCon Weekly – October 28, 2015 – Brian West Links: http://ift.tt/1Pa5q5e

1 0

Experience with SRTP
by Michael Nielsen 30 Oct '15

30 Oct '15

I'm running FS with TLS. This works. I would like to enable SRTP. Whenever I add that - either in vars.xml with rtp_secure_media=true or directly in my dialplan, then calls are dropped just after I'm answering them. I can still call subscribers, and the phone rings. When accepting the call it gets dropped with INCOMPATIBLE_DESTINATION. I have Bria on both ends.

2 1

Best way to disable 5060 and 8080
by Michael Nielsen 30 Oct '15

30 Oct '15

I'm only using FS with TLS, and do not need port 5060 and 5080 - I guess. I can see a lot of "bots" are trying to connect all the time - without luck. Should I disable port 5060 and 5080 so the load do to bots trying to make calls would disappear? And how do I make this best, via firewall or via FS (and how to I do it in FS, if that's the best way)?

4 6

Meaning of CF_AVPF
by Johannes Singler 30 Oct '15

30 Oct '15

What does it mean exactly when the CF_AVPF flag is set for a channel? Apparently, it was called CF_WEBRTC formerly, and indeed, it seems to be set when either ICE or media_webrtc are enable. However, the most specific thing about it is the "F", which stands for "early feedback", and that could be specified for non-WebRTC streams as well AFAIK (RFC4585). * Is is really set if and only if the profile type is AVPF? ** At least it seems to also be triggered by SAVPF. There are many checks that skip (SDES-based) SRTP setup if CF_AVPF is set, e.g. in switch_core_session_parse_crypto_prefs and switch_core_session_check_outgoing_crypto. Is this based on the fact that WebRTC uses DTLS and thus SDES should be disabled? IMHO, "Early feedback" should not trigger these decisions, though. Isn't it mostly the assumed implied DTLS that motivates those decisions? Shouldn't they rather be based on CF_DTLS or CF_DTLS_OK then? Should we maybe rename it back to CF_WEBRTC for some less confusion? And then there is also CF_AVPF_MOZ, which seems to work around Firefox particularities!? -- Johannes Singler Software Engineer Qnective Thurgauerstrasse 54 | 8050 Zürich | Switzerland www.qnective.com<http://www.qnective.com/> | johannes.singler(a)qnective.com<mailto:johannes.singler@qnective.com>

1 0

Intermittent Long Delay On Startup Centos 7.1
by David Wylie 30 Oct '15

30 Oct '15

I was running the following :Centos 7.1FreeSWITCH Version 1.7.0+git~20151021T165609Z~9fee9bc613~64bit (git9fee9bc 2015-10-21 16:56:09Z 64bit)Digital Ocean hosted "droplet", 2gig ram, 40 gig SSD, 2 CPU though it also happens with FreeSWITCH Version 1.6.2+git~20150930T182428Z~c0c3c38f64~64bit (git c0c3c38 2015-09-30 18:24:28Z 64bit) I have asked this on SO as well : http://stackoverflow.com/q/33305393/1642105 Intermittently, but about as often as 1 in 3, the start up process for Freeswitch pauses for a very long time (variable, but has been as long as 14 minutes). Running from the terminal window, here's the output. Notice the time between the "Add Task 2" and the next line :# /usr/bin/freeswitch -nonat -core -db /dev/shm -log /usr/local/freeswitch/log -conf /usr/local/freeswitch/conf -run /usr/local/freeswitch/run 2015-10-23 15:40:14.160101 [INFO] switch_event.c:685 Activate Eventing Engine. 2015-10-23 15:40:14.170805 [WARNING] switch_event.c:656 Create additional event dispatch thread 0 2015-10-23 15:40:14.272850 [INFO] switch_core_sqldb.c:3381 Opening DB 2015-10-23 15:40:14.282317 [INFO] switch_core_sqldb.c:1693 CORE Starting SQL thread. 2015-10-23 15:40:14.285266 [NOTICE] switch_scheduler.c:183 Starting task thread 2015-10-23 15:40:14.293743 [DEBUG] switch_scheduler.c:249 Added task 1 heartbeat (core) to run at 1445611214 2015-10-23 15:40:14.293837 [DEBUG] switch_scheduler.c:249 Added task 2 check_ip (core) to run at 1445611214 2015-10-23 15:49:47.883158 [NOTICE] switch_core.c:1386 Created ip list rfc6598.auto default (deny)If I spin up a virgin 6.7 machine (droplet) and follow my exact same install script, the longest pause I get at that spot is about 14 seconds, which is fine. Apart from the operating systems as supplied the two installations are similar. Does anyone have a guess as to why does it take so long on centos 7.1, and not on 6.7?And what is that check_ip doing? An amateurs look at the check_ip() in the source shows it resolving a hostname, but if I run a wireshark trace on all network interfaces it shows no activity at that point relating to DNS lookups or anything (it does if I ping google.com so I think I would have seen it). That's the same on 6.7 as well, so it's obviously not meant to, bit I'm not sure how else to determine what it's waiting for. Cheers.

8 8

mod_conference questions
by Dave Horton 30 Oct '15

30 Oct '15

I am trying to use mod_conference and have some questions (actually, a lot of questions — is there a forum/mailing list specifically devoted to conferencing? If so please let me know, that would be useful). I am using the (outbound) event socket, and I have two things that are puzzling me. The first is that when I send a ‘conference’ application request I get a CHANNEL_EXECUTE response from FSW but I never get a CHANNEL_EXECUTE_COMPLETE. The conference seems to be established fine, and I start getting conference::maintenance events, but I was expecting that I would get a CHANNEL_EXECUTE_COMPLETE event when the conference was established (or caller joined to existing conference). Is my expectation misplaced, or am I possibly doing something wrong? Secondly, I want the first call leg that I put into the conference to get all of the add-member (and other messages) from other call legs as they join and leave (this first leg I intend to use as a sort of “control leg” that will keep track of overall conference statistics and so forth). To do this, ater the first leg joins I issue this filter command: filter Conference-Unique-ID 5de34a49-4384-4ab4-9a26-4bb2a8610239 and I get back a success response. However, when I then connect a second caller to the conference, that first leg does not get the ‘add-member’ event (the _second_ leg does get this event). Am I doing something wrong? This was described here: https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket, and I am trying to follow that model but it is not working for me. Thanks in advance for any advice/direction/assistance. Dave

2 1

OPUS and bitrate
by Dragos Oancea 30 Oct '15

30 Oct '15

Hi Michael, Try a lower maxaveragebitrate together with : <param name="maxplaybackrate" value="8000"/> This will put Opus in Narrowband on Freeswitch side (FS encoder). If Bria is respecting the FMTP params that FS sends , it should send Narrowband voice too . I am not familiar with Bria , but if it shows 30-60 kb/s , perhaps it shows the "network bitrate" ( together with packet headers ).Note that maxaveragebitrate is the codec bitrate and actually Opus can go higher than that at times during the call but certainly not as high as 60 kb/s. Regards,Dragos

1 0

OPUS and bitrate
by Michael Nielsen 30 Oct '15

30 Oct '15

My users are primarily running on 3G networks and are experience some delays/lost packages. I would like to try and force a lower bitrate on the OPUS codec to see if it helps. They are using the Bria client on iOS. In my FS I've set this in opus.conf.xml: <param name="maxaveragebitrate" value="20000"/> But Bria are reporting an average between 30-60kb/s. Shouldn't the setting force a max of 20kb/s and are there more things to consider to get good voice quality on 3G networks and such? Package loss are reported from Bria to be around an average of 1-5%

1 0

Sending SMS to Mobile using FreeSWITCH mod_sms
by tarik＠pasifiktelekom.com.tr 29 Oct '15

29 Oct '15

2 1