June 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch Week in Review(Master Branch) June 15st-21st
by Kathleen King 25 Jun '14

25 Jun '14

Hello, again. This week in the FreeSWITCH master branch was quiet with only 10 commits. We had a couple of new changes with improvements to idle detection in mod_rayo and external ipv6 was added to the vanilla sip_profiles. The following bugs were squashed: 10719e5 work around macro definition of fileno breaking c++ code in mod_verto 2325981 FS-6593 removed unused 'struct ucred credentials' from mod_verto Jira: http://jira.freeswitch.org/browse/FS-6593 New features that were added: 1893c7b Added External ipv6 to vanilla sip_profiles 3c7c167 improved idle detection- call is idle only if no active components and not joined, in mod_rayo Improvements in cross platform build supports: 7b974de Add back mod_perl to debian build Changes to packaging: 062b7f1 Add mod_rtc and mod_verto to debian control-modules 9908476 Pull mod_verto from debian packaging temporarily 4d5153c Fix ordering in debian control-modules Miscellaneous commits: 13e0227 FS-6596 Use portable version of strerror_r() to fix a BSD compiler issue introduced in commit 12da85ef e7ee405 Add ldns to .gitignore Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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Core parameters questions
by Guillermo Ruiz Camauer 24 Jun '14

24 Jun '14

There are a few Core Parameters in switch.conf that I would like to know when they should be tweaked from their default configuration: max-db-handles: Should there be one handle every N simultaneous calls? What is N? initial-event-threads: Does it hurt or help to bump this up to 2 or 3 on an 8 core machine, even if the WARNING "Create additional event dispatch thread" hasn't come up? db-handle-timeout: What symptoms would indicate that this needs tuning? Should it somwhow be scaled when max-db-handles is changed?Thanks, -- Guillermo

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How to configure Freeswitch for SIP over WebSocket behind server-side NAT?
by Bernhard Wendel 24 Jun '14

24 Jun '14

Dear all, I have a FreeSWITCH server that lies behind a NAT on the server side. While usual SIP-Calls (using UDP) are working, I'm facing issues when it comes to the WebSocket Server. While specifying the external IP in the profile definition (<param name="ext-sip-ip" value="$${external_rtp_ip}"/>), on WebSocket the server sends the private instead of the public ip. The reverse proxying is done using nginx. As I didn't found an answer on the web, I hope you can help me! Here is a snippet of the ws communication (FreeSWITCH is answering the INVITE; 172.XX.42.1 is the private gateway and public nginx server; 172.XX.0.3 is the private IP the FreeSWITCH server uses): SIP/2.0 200 OK Via: SIP/2.0/WS ii6c2sdshbua.invalid;branch=z9hG4bK7579777;received=172.XX.42.1;rport=43567 From: <sip:13c0f2b0-b5b8-4822-afe6-38930cddc646@domain.tld>;tag=8a01krubf0 To: <sip:a3352c35-7a6d-4c9e-aeea-1a5f74e90b6f@domain.tld>;tag=p2K9pvtH70Nmm Call-ID: l6ijpvvic6l1ung778sr CSeq: 6213 INVITE Contact: <sip:a3352c35-7a6d-4c9e-aeea-1a5f74e90b6f@PUBLIC.IP:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.3.4-n20140511T185316Z-1~trusty+1+git~20140509T214701Z~7e0995d069~64bit [...] o=FreeSWITCH 1403358295 1403358296 IN IP4 *172.XX.0.3* s=FreeSWITCH c=IN IP4 *172.XX.0.3* [...] a=ice-pwd:6MXY3JdpOCdmDT9W a=candidate:0823320109 1 udp 659136 *172.XX.0.3* 25176 typ host generation 0 The nginx proxy configuration looks like the following: location /sip { access_log off; proxy_pass http://172.XX.0.3:5066; proxy_set_header X-Real-IP $remote_addr; proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for; # WebSocket support (nginx 1.4) proxy_http_version 1.1; proxy_set_header Upgrade $http_upgrade; proxy_set_header Connection "upgrade"; proxy_redirect off; } What configuration options do I have to set, so that FreeSWITCH uses the public IP for candiate creation and all other occurences instead of the private one? Or is it easier to adjust nginx configuration to replace the IP? Kind regards Bernhard

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Multi-Tenant Versus Multiple FS Instances
by Jerry Richards 24 Jun '14

24 Jun '14

Has anyone considered the Multi-tenant implementation (as described in the FS Wiki) versus having multiple Freeswitch instances running on the same machine (i.e. one instance for each company)? I am wondering if there is a trade-off between these two approaches? Regards, Jerry

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mod_xml_curl parse problem and solution
by Colin Martin 24 Jun '14

24 Jun '14

Hi, I was on IRC briefly yesterday asking about a parsing issue I had with mod_xml_curl. Essentially although the XML was correct, the parsing was failing. The person I was discussing it with suggested removing the header (<?xml version="1.0" encoding="UTF-8" standalone="no”?>) which was being generated by the library I was using to generate the XML. This header appears in many examples in books etc and so it had not occurred to me to remove it. Anyway, removing it worked and now mod_xml_curl is working perfectly for my needs, so thanks to the person who suggested it (sorry I can’t remember the IRC nick), and hopefully this’ll turn up in someone else’s Google search if they have a similar problem in the future. Thanks, Colin Martin

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Announcing FreeSWITCH Chinese Book and some FreeSWITCH updates from China
by Seven Du 24 Jun '14

24 Jun '14

Hi list, I’m glad to announce that the first Chinese book about FreeSWITCH (FreeSWITCH the Definitive Guide ) is available - http://book.dujinfang.com . During the years, the Chinese Community was growing and we have 1191 members in our local IRC Group - QQ. And we just finished a tiny ClueCon called FreeSWITCH Developers’s Salon where 100+ people joined us last week. We also suscessfully done 2 FreeSWITCH training sessions in Spring and Summer, Shanhai and Beijing respectively. Each has 30+ people attended and they came from everywhere in China mainland, and Taiwan. The FreeSWITCH Developers’s Salon 2014 is non profitable, but we got a few sponsors this year and a little money left so we donated to the FreeSWITCH community. ClueCon is coming and thanks Anthony and his team brought us so great software. I’m not doing Ads but China has biggest market in every aspects so if you are expand your services to China or finding VoIP terminations etc I might have some thoughts or suggestions. Meet at ClueCon. -- Seven Du http://about.me/dujinfang http://www.dujinfang.com http://www.freeswitch.org.cn

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Opus changes?
by Peter Villeneuve 24 Jun '14

24 Jun '14

I just realised that after updating to latest master, I can no longer negotiate Opus with CSipSimple. Before this latest update, it worked fine. I believe this maybe related to adding stereo but not sure. [DEBUG] switch_core_media.c:3469 Audio Codec Compare [opus:111:48000:20:0:2]/[opus:116:48000:20:0:1] [DEBUG] switch_core_media.c:3504 Audio Codec Compare [opus:116:48000:20:0:1] is saved as a near-match I have set "global_codec_prefs=OPUS,G722,iLBC,SILK,PCMU,PCMA,GSM" What do I have to change to have Opus available again? Thanks

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ZRTP negotiation taking almost 10 seconds every time
by Peter Villeneuve 24 Jun '14

24 Jun '14

Hi, I've been trying to debug why ZRTP negotiation with FS as endpoint takes almost 10 seconds every single time. I remember in the past ZRTP negotiation used to take 2 to 3 seconds at most. I'm calling MoH extension from CSipSimple with ZRTP enabled to test it out and indeed it always takes 9 seconds to negotiate the ZRTP exchange. And even after it does it successfully and enteres secured state, I still see a lot of errors related to failed decryption. I have posted the FS and CSipSimple logs on pastebin. FS log here http://pastebin.com/WNGcpTqB and CSipSimple log here http://pastebin.com/DfW5gqSW These are the relevant excerpts from FS which may help to shed light as to what's going on. You can see that the zrtp hellos from FS apparently never reach the CSipSimple client: *2014-06-10 16:17:55.949864 [DEBUG] switch_rtp.c:1352 [ zrtp engine]: WARNING! HELLO Max retransmissions count reached (20 retries). ID=9* But then you see a five second interval of nothing until CSipSimple's ZRTP hello reaches FS and the negotiation starts: *2014-06-10 16:17:55.949864 [DEBUG] switch_rtp.c:1352 [ zrtp]: Stream ID=9 CLEAR switching <START> ---> <NOZRTP>.* *2014-06-10 16:18:00.789889 [DEBUG] switch_rtp.c:1352 [ zrtp utils]: Received <Hello > packet with ssrc=0 seq=1/1 size=168. Stream9:CLEAR:NOZRTP.* Once that's successfull indeed ZRTP appears to work properly, although I still see a lot of errors like this: 2*014-06-10 16:18:01.509898 [DEBUG] switch_rtp.c:1352 [zrtp protoco]: ERROR! Decrypt failed. ID=9:DH s=SRTP authentication failure (RTP size=51 ssrc=690666908 seq=20192/20192 pt=111)* *2014-06-10 16:18:01.509898 [DEBUG] switch_rtp.c:1352 [ zrtp engine]: Add 140 bytes of entropy to the RNG pool.* Now I guess this could be due to dropped packets, but I'm curious if others have seen the same thing in their setups. Also, I see this message in the FS logs. On the CS logs everything seems to be fine. Why does FS think that CS needs new un-enrolment? I presume that nedpint is actually a typo for endpoint, but this just made me more confused. *2014-06-10 16:18:01.769867 [DEBUG] switch_rtp.c:1352 [zrtp protoco]: INFO! The user requires new un-enrolment - the nedpint may clear the cache or perform other action. ID=9* *c60c4480-f0ba-11e3-aa6e-ddba267b1226 2014-06-10 16:18:01.769867 [DEBUG] switch_rtp.c:1323 User unenrolled!* *2014-06-10 16:18:01.769867 [DEBUG] switch_rtp.c:1352 [ zrtp cache]: cache_put() zid1=346638366666393864366363, zis2=3888449c219de624b44b3ece MiTM=NO* *2014-06-10 16:18:01.769867 [DEBUG] switch_rtp.c:1352 [ zrtp cache]: cache_put() Just update existing value.* Any ideas as to what's happening? Thanks

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question regarding xml_handler in FreeSWITCH/FusionPBX OpenSIPS integration
by Saeed Abbassi 24 Jun '14

24 Jun '14

> Hello all, > > > Regarding the integration of FreeSWITCH with OpenSIPS, > Is it necessary to have the xml_handler.lua, although fusionpbx is installed and has its own xml_handler/index.lua ? > > As you have guessed, I have a FreeSWITCH/FusionPBX setup, trying to integrate it with OpenSIPS. > > Any comment would be much appreciated, > > Best regards, > Saeed A

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mod_verto INCOMPATIBLE_DESTINATION
by Ciprian Dosoftei 24 Jun '14

24 Jun '14

Hello, I am trialling the new mod_verto endpoint and so far it's been fantastic, definitely a great addition! I am trying to figure out what I am doing wrong in terms of codec settingss as I can only get it working with opus, PCMU will simply throw INCOMPATIBLE_DESTINATION if the channel is answered. On the very same system I don't have any issues getting PCMU over webRTC by using sofia's own WSS binding. Below are my codec string settings for verto (dropping opus will result in failure): <param name="outbound-codec-string" value="opus,PCMU,PCMA"/> <param name="inbound-codec-string" value="opus,PCMU,PCMA"/> Thank you! -- Best Regards, Ciprian Dosoftei The information transmitted is intended only for the addressee and may contain privileged and/or confidential material. If you are not the intended recipient, kindly contact the sender and delete the message. Any disclosure, distribution or copying of this message is strictly prohibited without the expressed permission of the sender.

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