April 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

webrtc: [ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP
by Quan Huo Sheng 12 Apr '16

12 Apr '16

Hi All; I want to use freeswitch to set up a WebRTC POC (Audio only, udp/tls/rtp/savp). Setup is Anonymous (192.168.199.216/chrome/sipjs:sip-0.7.0.min) ->freeswitch (192.168.199.22) ->1010 (192.168.199.216/chrome/sip-0.7.0.min),just following the information in book "FreeSWITCH 1.6 Cookbook". If using media bypass mode (inbound-bypass-media == true), all works fine, caller and called can hear each other. But if disabling media bypass mode, call is rejected by FS. FS log shows "[ERR] mod_sofia.c:2299 CODEC NEGOTIATION ERROR. SDP". Attachment is detailed FS log file. Chrome uses opus 111, FS uses opus 116. Mod_sofia.c::sofia_receive_message() --> sofia_media.c::sofia_media_negotiate_sdp() --> sofia_core_media.c::sofia_core_media_negotiate_sdp(). Help is needed to troubleshoot this issue. Thanks advance. Smile. [This e-mail is confidential and may be privileged. If you are not the intended recipient, please kindly notify us immediately and delete the message from your system; please do not copy or use it for any purpose, nor disclose its contents to any other person. Thank you.] ---ST Electronics Group---

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Codec renegotiation on Re-INVITE withot SDP
by Станислав Запольский 12 Apr '16

12 Apr '16

Hello. We are facing the following problem with our FS setup: 1. provider sends us INVITE with "a=sendonly" in SDP 2. FS picks up the call, playing ivr and replying with 200 and "a=recvonly" in SDP 3. provider sends Re-INVITE without SDP 4. FS replying 200 with already negotiated SDP (single codec and "a=recvonly") Provider wants renegotiation of sdp on step 4, or a least "a=sendrecv" parameter. Is there a way to achieve this without source modification? We tried setting following profile parameters: renegotiate-codec-on-hold=true renegotiate-codec-on-reinvite=true enable-3pcc=true but no luck. The only solution we found was modifing sofia.c like this: line 7374 - replace switch_core_media_gen_local_sdp(session, SDP_TYPE_RESPONSE, NULL, 0, NULL, 0); with switch_core_media_gen_local_sdp(session, SDP_TYPE_RESPONSE, NULL, 0, "sendrecv", 0); line 7376 - replace nua_respond(tech_pvt->nh, SIP_200_OK, TAG_END()); with nua_respond(tech_pvt->nh, SIP_200_OK, SOATAG_USER_SDP_STR(tech_pvt->mparams.local_sdp_str), TAG_END()); Thanks.

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Bounty
by Abaci B 11 Apr '16

11 Apr '16

Looking for developer that can help with a feature development, I currently have a $200 bounty on it, if anyone is interested please let me know. https://freeswitch.org/jira/browse/FS-8700

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Re: [Freeswitch-users] nibblebill do not terminate calls properly
by jelena＠misticnabica.hr 11 Apr '16

11 Apr '16

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Thursday FreeSWITCH Bug Hunt
by Ítalo Rossi 11 Apr '16

11 Apr '16

FreeSWITCHers, Join us thursday 2PM CST for the Thursday FreeSWITCH Bug Hunt! Where? [conference.freeswitch.org/vc/#/?autocall=888](https://conference.frees witch.org/vc/#/?autocall=888 "https://conference.freeswitch.org/vc/#/?autocall=888" ) Chat? <https://hipchat.freeswitch.org/gYzPMg3Oc> What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support! Help us help you, Join the Bug Hunt! Ítalo Rossi italo(a)freeswitch.org IRC chat.freenode.net #freeswitch #freeswitch-dev Bugs? https://freeswitch.org/jira Docs? https://freeswitch.org/jira Chat? https://hipchat.freeswitch.org/gUdAgy0m6

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NATIVE SQL ERR [cannot commit - no transaction is active]
by Chandramouli P 11 Apr '16

11 Apr '16

Hi, Please find my below deployed environment: Environment: Microsoft Azure OS: CentOS 7.0 (64 bit) FreeSwitch Version: 1.6.6~64bit ( 64bit) I installed FreeSwitch through "Yum" and works fine for couple of hours. After that I am getting below errors: [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [attempt to write a readonly database] BEGIN EXCLUSIVE 2016-04-05 13:29:33.368355 [CRIT] switch_core_sqldb.c:1952 ERROR [attempt to write a readonly database] 2016-04-05 13:29:33.368355 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [cannot commit - no transaction is active] Can anybody tell me what could be the issue and how to solve this? Thanks in advance, Chandra.

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question on handling of nonce count (nc)
by Dave Horton 11 Apr '16

11 Apr '16

In investigating some REGISTER storms on one of my networks, I am seeing some client devices interacting with Freeswitch in a manner that can lead to excessive registration traffic. It looks to me to be more of an endpoint problem than a freeswitch problem, but I would like confirmation of that as well as any ideas on how to handle this (i.e., throttle back this traffic). The basic problem flow is this: - Client sends a REGISTER with a large nc value and nonce value A - Freeswitch replies 401 with stale=true (nonce is stale) and nonce value B - Client sends another REGISTER with nc value incremented by 1 and nonce value A again - Freeswitch replies 401 with stale=true (nonce is stale) and nonce value C - Client sends another REGISTER with nc value incremented again and nonce value A again ….etc. This seems particularly problematic with some Yealink, Communicator, and Polycomm IP Soundlink endpoint Here is a specific example (some information redacted) recv 804 bytes from udp/[]:5060 at 23:54:11.906859: ------------------------------------------------------------------------ REGISTER sip:x.x.x.x:5060 SIP/2.0 Authorization: Digest username="123371",realm="sip.foo.com",nonce="41adc443-57c8-4325-831e-ffd006a922d4",uri=“sip:x.x.x.x:6060",response="3a4b5f05ec1897a58865b4ba0cdb0b4d",cnonce="b5d06adf6a4c7c0592f5fc1d7766a605",nc=0000008a,qop=auth,algorithm=MD5 send 641 bytes to udp/[10.128.77.170]:5060 at 23:54:11.909722: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized WWW-Authenticate: Digest realm=sip.foo.com", nonce="888c8919-b28f-4be4-be12-753430aafa88", stale=true, algorithm=MD5, qop=“auth” recv 804 bytes from udp/[]:5060 at 23:54:12.007622: ------------------------------------------------------------------------ REGISTER sip:x.x.x.x:5060 SIP/2.0 Authorization: Digest username="123371",realm="sip.foo.com",nonce="41adc443-57c8-4325-831e-ffd006a922d4",uri=“sip:x.x.x.x:6060",response="556498e38d27c944f10e3a0c11a5ea41",cnonce="5585e516afcf2f95bfbc4bef11a075ee",nc=0000008b,qop=auth,algorithm=MD5 send 641 bytes to udp/[10.128.77.170]:5060 at 23:54:12.010376: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized WWW-Authenticate: Digest realm=“sip.foo.com", nonce="1ff3b9a3-4cbb-4569-b6c7-7bee203547ac", stale=true, algorithm=MD5, qop="auth" recv 804 bytes from udp/[10.128.77.170]:5060 at 23:54:12.108742: ------------------------------------------------------------------------ REGISTER sip:x.x.x.x:5060 SIP/2.0 Authorization: Digest username="123371",realm="sip.foo.com",nonce="41adc443-57c8-4325-831e-ffd006a922d4",uri=“sip:x.x.x.x:6060",response="9cf2360ef5f28684e667ac878362d0c0",cnonce="9833d8d3889d3ae8875e0f6f00c4d3f3",nc=0000008c,qop=auth,algorithm=MD5

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FreeSWITCH malforming SIP Package
by Nagorny, Dimitry 11 Apr '16

11 Apr '16

Good afternoon Ladies and Gentlemen, I'm using FreeSWITCH (1.7.0 git cae2553 2015-12-11 00:23:48Z 64bit) as B2BUA between Kamailio and our own Server. Our Server is attached to FreeSWICH via ESL. >From Kamailio into our own Server it is connecting w/o problems. But when I initiate a call from our Server to Kamailio via FreeSWITCH I've a weird reaction: Network-Overview: PSTN(192.168.1.30) --> Kamailio(192.168.1.150/10.250.5.74) --> FreeSWITCH(10.250.5.72) --> Our Server(10.250.5.67) 1st Initial Invite to FS: INVITE sip:7@10.250.5.72 SIP/2.0 Via: SIP/2.0/UDP 10.250.5.17:5090;branch=z9hG4bK.GSRDySx-y;rport From: <sip:1001@10.250.5.72>;tag=nDaJDf3lz To: sip:7@10.250.5.72 Route: <sip:10.250.5.72:5080;lr> Content-Type: application/sdp 7 is mapped to go to our Server and our Server then is trying to connect to a UA (2031) at PSTN via FreeSWITCH gateway(Kamailio): INVITE sip:2031@10.250.5.74:5070 SIP/2.0 Via: SIP/2.0/UDP 10.250.5.72:5070;rport;branch=z9hG4bKmgrc4De2Q0ySN From: "" <sip:1001@[10.250.5.72:5070]>;tag=U5jj715ZB6maK To: <sip:2031@10.250.5.74:5070> This Invite was built by FreeSWITCH and I could confirm that the "SIP-Invite" sent via ESL to FreeSWITCH from our Server was correct but as you can see here FreeSWITCH "malformed" the From-Header with the square brackets and due to that I'm getting error from PSTN later: SIP/2.0 400 Bad Request Warning: 399 192.168.1.30 "Malformed headers : From " To: <sip:2031@10.250.5.74:5070>;tag=ae0f6cb7ca85c1fb958bd8bed1f9b7b7 Here some config data that is set in FreeSWITCH: Vars.xml: <X-PRE-PROCESS cmd="set" data="domain=10.250.5.72"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5070"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5080"/> Sip-profiles/external/testpbx.xml: <include> <gateway name="testpbx"> <param name="realm" value="10.250.5.74:5070"/> <param name="from-domain" value="10.250.5.72:5070"/> <param name="proxy" value="10.250.5.74:5070"/> ... </gateway> </include> Sip-profiles/external.xml and internal.xml: <param name="rtp-ip" value="10.250.5.72"/> <param name="sip-ip" value="10.250.5.72"/> <param name="ext-rtp-ip" value="10.250.5.72"/> <param name="ext-sip-ip" value="10.250.5.72"/> Can anyone help why FreeSWITCH is doing this or can point to a class where the the From Header is built? Maybe someone knows a solution? Here you can find full SIP-Trace Logs and Full FreeSWITCH Debugging Log: https://we.tl/FtPuknRrSw Very Respectfully Dimitry Nagorny Trainee

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Configure SDP
by amani mansour 11 Apr '16

11 Apr '16

Good morning , I need to configure a number to send invite without SDP , can anyone help me please ? best regards amani

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Best way to record each leg of a call separately?
by Piotr Starzyk 11 Apr '16

11 Apr '16

I’m trying to record phone calls, and phone conferences, so that later on their audio can be reviewed. The requirement is for each leg of the call to be recorded separately. After doing some googling, one way of doing it, is by using record_session and setting RECORD_STEREO to true. That will result in caller and receiver audio streams being placed in separate channels. At that point you could split the channels to get the individual streams. Is that the recommended approach, or is there a better way (or module) to record legs separately? Also, am I correct assuming the recording is done on FreeSWITCH server? Does that mean that each SIP client will need to send two audio streams out (one to the other client and one to freeswitch for recording?). Thanks.

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