June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] SIP 500 overlapping requests due to race condition
by Phil Quesinberry 23 Jun '14

23 Jun '14

Digging into this further, I found the following document detailing best current practices for handling this condition. Here is a clip of the relevant section from RFC 5407 - Example Call Flows of Race Conditions in the Session Initiation Protocol (SIP) http://tools.ietf.org/html/rfc5407 Hasebe, et al. Best Current Practice [Page 16] RFC 5407 Example Call Flows of Race Conditions December 2008 3.1.4. Callee Receives re-INVITE (Established State) While in the Moratorium State (Case 1) State Alice Bob State | | | ini-INVITE w/offer1 F1 | |------------------------------->| Pre | 180 F2 | Pre |<-------------------------------| Ear | | Ear | 200(ini-INV) w/answer1 F3 | |<-------------------------------| Mora | ACK F4(packet loss) | Mora |-------------------->x | Est | | | re-INVITE F6 200 F5(=F3) | | w/offer2 w/answer1 | |------------- --------------| | \ / | | X | | / \ | |<------------ ------------->| *race* | 200(re-INV) F8| | ACK F7(=F4) w/answer2 | |------------- --------------| | \ / | | X | | / \ | |<------------ ------------->| | ACK (re-INV) F9 | Est |------------------------------->| | | | | This scenario illustrates the race condition that occurs when a UAS in the Moratorium state receives a re-INVITE sent by a UAC in the Established state. The UAS receives a re-INVITE (with offer2) before receiving an ACK for the ini-INVITE (with offer1). The UAS sends a 200 OK (with answer2) to the re-INVITE (F8) because it has sent a 200 OK (with answer1) to the ini-INVITE (F3, F5) and the dialog has already been established. (Because F5 is a retransmission of F3, SDP negotiation is not performed here.) As can be seen in Section 3.3.2 below, the 491 response seems to be closely related to session establishment, even in cases other than INVITE crossover. This example recommends that 200 be sent instead Hasebe, et al. Best Current Practice [Page 17] RFC 5407 Example Call Flows of Race Conditions December 2008 of 491 because it does not have an influence on the session. However, a 491 response can also lead to the same outcome, so either response can be used. Moreover, if the UAS doesn't receive an ACK for a long time, it should send a BYE and terminate the dialog. Note that ACK F7 has the same CSeq number as ini-INVITE F1 (see Section 13.2.2.4 of RFC 3261 [1]). The UA should not reject or drop the ACK on grounds of the CSeq number. _____________________________________________ From: Phil Quesinberry Sent: Friday, June 20, 2014 3:49 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: SIP 500 overlapping requests Correction: I meant to say "if FreeSWITCH could be configured to either wait up to x milliseconds for an ACKNOWLEDGEMENT to arrive." ___________________________________ From: Phil Quesinberry Sent: Friday, June 20, 2014 3:36 PM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: SIP 500 overlapping requests As it turns out, my assumption was correct. We were not receiving an acknowledge before the second invite from the carrier. One of their support engineers determined that they had recently enabled acknowledgement logging on their proxy/SBC, which apparently delayed transmission of the acknowledgement by a millisecond or two, which was enough to cause the invite from the server handling the media/call to arrive just slightly before the acknowledgement, so FreeSWITCH would tear down the call. The problem was intermittent since the acknowledgement would sometimes arrive first. They disabled acknowledgement logging and the problem went away. With this and UDP in mind, it would likely reduce the incidence of dropped calls if FreeSWITCH could be configured to either wait up to x milliseconds for an invite to arrive, or allow x number of overlapping requests before throwing an error. I saw a post where Tony mentioned that a patch had already been crafted to allow overlapping requests in the past, so it seems that most of the work has been done. __________________________________ From: Phil Quesinberry Sent: Thursday, June 19, 2014 11:08 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: SIP 500 overlappting requests We're having intermittent problems receiving calls from a particular DID. The basic scenario is as follows: Carrier sends invite w/SDP We respond w/OK and SDP when call is answered Carrier sends another invite with incremented sequence number and updated SDP (aren't they supposed to ACK/respond before doing this?) We send SIP 500 - overlapping requests Carrier sends BYE - terminating the call Sometimes (rarely) the call completes, most of the time FS sends a 500 and doesn't get the call. This just started happening recently and does not occur with any other DIDs, including other DIDs from the same carrier. I updated to the latest stable GIT to see if that might solve the problem but I'm pretty sure this is on the carrier's end. They're trying to blame it on us since we're sending the 500 and suggested that we allow overlapping requests. I'm assuming that their end needs to acknowledge our answer OK/SDP before sending their own but if that's the problem then it raises another question in my mind: since the SIP transport is generally UDP, what if we don't receive their ACK before receiving their subsequent invite/SDP? This could certainly happen and would cause the call to be rejected if the above is true. I updated to the latest stable GIT last night to rule that out as a possible cause of the problem. Is there a way to get FS to allow a limited number of overlapping requests as a possible workaround? If someone thinks this looks like it could be an FS bug, I'll file a Jira. Here is a trace which shows this happening: http://pastebin.freeswitch.org/22769 Thanks, Phil Quesinberry Q Systems Engineering, Inc. Embedded Hardware/Software Development (410) 969-8002 http://www.qsystemsengineering.com

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Cluscon Schedule is Filling Up! You don¹t want to miss this one!
by Ken Rice 23 Jun '14

23 Jun '14

ClueCon is rapidly approaching! The schedule is rapidly filling and this year will be better then ever! Join us in Chicago August 4th 7th! Here¹s just a few of the people you¹ll see! Simon Woodward, CEO of Simwood ( http://simwood.com ) and noted VoIP Fraud Researcher John Riordan, Cofounder and CTO of OnSip ( http://onsip.com ) Bogdan-Andrei Iancu, Founder and Lead Developer of OpenSIPS ( http://opensips.org ) James Body, Head of R&D at Truphone ( http://truphone.com ) Emil Ivov, Project Leader, Jitsi.org Ben Klang, Technology Strategist at Mojo Lingo and Project Lead for Adhearsion ( http://www.adhearsion.com/ )` Darren Schreiber, CEO 2600Hz ( http://2600hz.org/ ) This is just a small sample of the Speakers this year! Register at ClueCon.com now and get extra entries into the door prize drawings while you still can! The year we have a pile of goodies from our friends at ThinkGeek ( http://www.thinkgeek.com ) including a Laser-Guided Tactical Tie, Gunnar Intercept Computer Glasses, the Thing Geek Annoy-A-Tron and others! See https://www.cluecon.com/thinkgeek-sponsors-cluecon-giveaways/ for more information!. Don¹t forget ClueCon starts Monday Afternoon this year! Monday will be the Hack-A-Thon will all the developers around! And Stay later in the Week for FreeSWITCH Training! Sign up for the training now as space is limit! See https://www.cluecon.com/freeswitch-training-is-coming-to-cluecon/ for more information! And for the hacker/maker in you, Truphone will be sponsoring Dangerous Demos! Show off your latest creation in a 3 to 4 minute Demo and the best demos will win a prize! Audience Live Voting will determine the winners! Visit https://www.cluecon.com/dangerous-demos-coming-to-cluecon/ for more information and entry details! ClueCon 2014 is shaping up to be the absolute best ClueCon Ever! Don¹t miss out on the Open Source VoIP Conference that is For Developers and organized by developers! Follow us on Twitter @ClueCon for updates as they happen! See you at ClueCon! Https://www.cluecon.com/ -- Ken http://www.ClueCon.com http://www.FreeSWITCH.org http://www.OSTAG.org Twitter: @ClueCon http://www.FreeSWITCH.org

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secure websocket connection refused
by ik 23 Jun '14

23 Jun '14

Hello, I'm using Freeswitch 1.4.6 on CentOS 6 . I'ved configured Freeswitch to work with webrtc, and created certification based on http://wiki.freeswitch.org/wiki/SIP_TLS. I have also enabled the ws-binding and wss-binding at internal.xml. When doing netstat, I can see both 5066 and 7433 binded: netstat -lnpt | grep freeswitch tcp 0 0 37.139.XXX.XXX:7443 0.0.0.0:* LISTEN 27937/./freeswitch When I try to access the insecure ws (5066), I have a connection: tport.c:2749 tport_wakeup_pri() tport_wakeup_pri(0x7f095c00b530): events IN tport.c:862 tport_alloc_secondary() tport_alloc_secondary(0x7f095c00b530): new secondary tport 0x7f095c09b6a0 tport.c:2620 tport_accept() tport_accept(0x7f095c00b530): incoming secondary on ws/37.139.XXX.XXX:5066/sip failed. reason = WS_INIT But when I try to use the wss connection, I'm getting connection refused. I'm testing it on the same machine with the following command: openssl s_client -connect 37.139.XXX.XXX:7433 My cert folder has the following certifications: drwxr-x--- 2 root root 4096 Jun 23 08:42 CA -rw-r----- 1 root root 1270 Jun 23 08:42 agent.pem -rw-r----- 1 root root 729 Jun 23 08:42 cafile.pem -rw-r--r-- 1 root root 1270 Jun 23 08:46 dtls-srtp.pem -rw-r--r-- 1 root root 1270 Jun 23 08:46 wss.pem When using wireshark, I can see that the ACK packet get a reset flag after sending a SYN request. What am I missing here or doing wrong ? Thanks, Ido

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Shorewall QoS/EF Packet Marking
by Jason Moran 23 Jun '14

23 Jun '14

I have found the wiki entry for QoS to mark the packets with "EF" such that my ISP prioritizes the voice traffic accordingly: https://wiki.freeswitch.org/wiki/QoS However, we are using Shorewall (Shoreline Firewall) to manage IPTables on our CentOS 5.x server, and frankly this entry on Packet Marking in Shorewall is confusing: http://shorewall.net/PacketMarking.html Has anybody managed to implement this (from the FS QOS page) in Shorewall? : # Mark RTP packets with EF: iptables -t mangle -A OUTPUT -p udp -m udp --sport 16384:32768 -j DSCP --set-dscp-class ef Thanks, Jason

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setup presence for multiple FS
by Assaf Dahary 23 Jun '14

23 Jun '14

Hi, I have several FS servers in different locations and they are all managed from a central backend server DB. Currently each FS maintains its own local MySql DB tables which were created on installation. Routing calls between agents on different FS server are done with my own custom scripts. *I'm not using central sip proxy server like OpenSips/Kamailio. Now I would like to share presence between all agents that are registered on different FS servers, BUT I cannot make it because each FS server manages its own internal MySql DB with local presence. How can I share presence between all agents from all FS servers? Should I force all FS servers to share a single MySql DB on the backend server (set each DSN to point on the same remote DB)? Please advise Regards Assaf

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NDLB-connectile-dysfunction and 1.5.13b
by Iskren Hadzhinedev 23 Jun '14

23 Jun '14

Greetings! I'm looking at migrating my old FreeSWITCH 1.2.23 to 1.5.13b from git. I have a Yealink T42G phone that has lots of issues, one of them being call drops after 32 seconds. With 1.2.23 I was able to solve the problem using the following in my dialplan: <extension name="Yealink Detection" continue="true"> <condition field="${sip_user_agent}" expression="^Yealink"> <action application="set" data="sip-force-contact=NDLB-connectile-dysfunction"/> </condition> </extension> After the 1.5.13b upgrade this no longer works. The full version I'm currently using is: FreeSWITCH Version 1.5.13b+git~20140621T003222Z~47891d5caa~64bit (git 47891d5 2014-06-21 00:32:22Z 64bit) Is this intended or I'm missing something obvious here :) Thanks! Kind regards, -- Iskren Hadzhinedev

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any way to use a hook/event for voicemail notification instead of email?
by Nathan Neulinger 23 Jun '14

23 Jun '14

Seems like it would enable a bit more functionality if the notification were optionally handled as a hook, with the msg id/other data as parameters. In addition to enabling a bit more control of the content beyond the basic templates - this might also enable other functionality - such as triggering automatic transcription. I didn't see anything in the mod_voicemail wiki page, but figured I'd ask if there was any other trick for accomplishing this. From a look over the code, I'm not seeing any obvious way to have the event based codepath trigger but not the email notification unless I've missed something. A minimal means to handle this might be adding a parameter to disable voicemail email notification, but leave in all of the voicemail event generation. Thoughts? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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FreeSwitch running on VM. What all ports need to be forwarded.
by Anurag Rana 23 Jun '14

23 Jun '14

Hi all, I am running FS on a Virtual Machine which in turn is running on Windows server. I want to know what all ports I need to forward from Windows server to VM apart from 1. Sip Port - 5060 2. WebRTC Port - 9066 [ Reason for asking this question <https://jira.freeswitch.org/browse/FS-6618?filter=-2> ] -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences.

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Getting issue while using mod_rtmp
by Akib Sayyed 23 Jun '14

23 Jun '14

guys I am using mod_rtmp in freeswitch I am using flex client from freeswitch git repo. I am able to connect ,login place a call but not able to hear any voice or call is getting disconnected even if context is default it not routing properly here is cli log http://pastebin.com/DvaXECdl -- Akib Sayyed Matrix-Shell akibsayyed(a)gmail.com akibsayyed(a)matrixshell.com Mob:- +91-966-514-2243

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Emulate typical house phone/answering machine
by d-OP 23 Jun '14

23 Jun '14

Is there a way to configure freeswitch to allow a callee to screen a call by eavesdropping on the caller leaving a voicemail and then, if desired, pick up the handset to accept the call/pull it back from voicemail? This was discussed briefly here in Aug of 2012 but there was no resolution. Anybody have a working example they can share? Thanks, -Terry iPhone says Hello World!

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