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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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Call is interrupted in 30 seconds, configuring NAT on Amazon EC2
by Стас Тельнов 11 Apr '16

11 Apr '16

Hello. When using a call or conference through sip — freeswitch with external provider there is a problem – the call is interrupted in 30 seconds. Though the sound goes all right. I think that it caused by the NAT settings for freeswitch, but I don't understand how to adjust it correctly. At start of freeswitch I see the following mistakes in the tracking data: 2016-04-08 07:04:50.903529 [INFO] switch_nat.c:417 Scanning for NAT 2016-04-08 07:04:50.903678 [DEBUG] switch_nat.c:170 Checking for PMP 1/5 2016-04-08 07:04:51.903843 [DEBUG] switch_nat.c:170 Checking for PMP 2/5 2016-04-08 07:04:52.904023 [DEBUG] switch_nat.c:170 Checking for PMP 3/5 2016-04-08 07:04:53.904185 [DEBUG] switch_nat.c:170 Checking for PMP 4/5 2016-04-08 07:04:54.904360 [DEBUG] switch_nat.c:170 Checking for PMP 5/5 2016-04-08 07:04:55.904495 [ERR] switch_nat.c:199 Error checking for PMP [general error] 2016-04-08 07:04:55.904548 [DEBUG] switch_nat.c:422 Checking for UPnP 2016-04-08 07:05:07.905219 [INFO] switch_nat.c:438 No PMP or UPnP NAT devices detected! Despite of this mistake, conference communication between two internal users works normally. The problem arises at a call through external provider. We have the following architecture: In a cloud of Amazon EC2 there are 2 servers – opensips and freeswitch, both for NAT for external clients, but have an opportunity to work with each other directly. opensips has the internal address 172.31.0.169 and external 52. *.*.177 freeswitch has the internal address 172.31.22.124 and external 52. *.*.198 In fact, freeswitch acts only for conferences, and is ready for use of a remote DB on opensips. The auto-nat settings by default didn't work. The problem is in the external profile settings as far as I understand. I have filled and created the following configuration: vars.xml <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto”/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=52.*.*.198”/> <!— public freeswitch ip —> <X-PRE-PROCESS cmd="set" data="external_sip_ip=52.*.*.198”/> <!— public freeswitch ip —>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/tls"/> sip_profile/external.xml <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value=“52.*.*.198”/> <!— public freeswitch ip —> <param name="ext-sip-ip" value=“52.*.*.198”/> <!— public freeswitch ip —> In this sip_profile/external.xml I tried to fill rtp-ip/sip-ip and ext-rtp-ip/ext-sip-ip with the corresponding addresses of opensips server (that would be logical), but in that case conferences didn't work at all and errors below appeared: [ERR] sofia.c:2935 Error Creating SIP UA for profile: external ... Also I tried to put such configuration: <param name="rtp-ip" value="auto"/> <param name="sip-ip" value="52.*.*.198”/> but it also hasn't helped to solve the problem. autoload_configs/switch.conf.xml <param name="rtp-start-port" value="16384"/> <param name="rtp-end-port" value="32768"/> "sofia status" looks as follows: Name Type Data State ================================================================================================= 172.31.22.124 alias internal ALIASED external profile sip:mod_sofia@52.*.*.198:5060 RUNNING (0) external profile sip:mod_sofia@52.*.*.198:5061 RUNNING (0) (TLS) external::*********.com gateway sip:USER@*********.com REGED internal profile sip:mod_sofia@52.*.*.198:5080 RUNNING (0) internal profile sip:mod_sofia@52.*.*.198:5081 RUNNING (0) (TLS) ================================================================================================= 2 profiles 1 alias "sofia status profile external" looks as follows: ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 172.31.22.124 Ext-RTP-IP 52.*.*.198 SIP-IP 172.31.22.124 Ext-SIP-IP 52.*.*.198 URL sip:mod_sofia@52.*.*.198:5060 BIND-URL sip:mod_sofia@52. *.*.198:5060;maddr=172.31.22.124;transport=udp,tcp TLS-URL sip:mod_sofia@52.*.*.198:5061 TLS-BIND-URL sips:mod_sofia@52. *.*.198:5061;maddr=172.31.22.124;transport=tls HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA CODECS OUT PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 0 What do I adjust wrong? Whether there is some opportunity, to tell freeswitch not to break off a call in 30 seconds even if NAT isn't adjusted?

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Re: [Freeswitch-users] FreeSWITCH library for SRTP/DTLS
by jelena＠misticnabica.hr 10 Apr '16

10 Apr '16

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freeswitch with postgres setup process?
by jungle Boogie 10 Apr '16

10 Apr '16

Hello All, I have freeswitch installed from source on Debian 8.3 64 bit and Postgresql 9.4 installed from the Postgres packages[1]. What's the best way to have Freeswitch use a database that I'll setup and create? Do I need to recompile freeswitch with certain modules enabled? Do I need to enable to config flags? I've completed these things: install unixodbc-dev install odbc-postgresql update /etc/odbc.ini: [freeswitch] ; WARNING: The old psql odbc driver psqlodbc.so is now renamed psqlodbcw.so ; in version 08.x. Note that the library can also be installed under an other ; path than /usr/local/lib/ following your installation. Driver = /usr/lib/x86_64-linux-gnu/odbc/psqlodbcw.so Description=Connection to LDAP/POSTGRESQL Servername=localhost Port=5432 Protocol=6.4 FetchBufferSize=99 Username=freeswitch Password=xxx Database=freeswitch ReadOnly=no Debug=1 CommLog=1 view etc/odbcinst.ini [PostgreSQL ANSI] Description=PostgreSQL ODBC driver (ANSI version) Driver=psqlodbca.so Setup=libodbcpsqlS.so Debug=0 CommLog=1 UsageCount=1 [PostgreSQL Unicode] Description=PostgreSQL ODBC driver (Unicode version) Driver=psqlodbcw.so Setup=libodbcpsqlS.so Debug=0 CommLog=1 UsageCount=1 And finally, test the connection: # isql -v freeswitch +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ [1] http://www.postgresql.org/download/linux/debian/ -- ------- inum: 883510009027723 sip: jungleboogie(a)sip2sip.info xmpp: jungle-boogie(a)jit.si

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Re: [Freeswitch-users] freeswitch with postgres setup process?
by jelena＠misticnabica.hr 10 Apr '16

10 Apr '16

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FreeSWITCH in cluster
by Dmitry Mordovin 10 Apr '16

10 Apr '16

Hi, I want configure two FreeSWITCH servers as cluster. As I understand, I need: - Configure one core DB (PostgreSQL) for each FreeSWITCH instances. - Add two A records in DNS to resolve domain name and load balancing. *Scheme, 2 FS servers and one DB.* FS-1 server 192.168.0.1 Core-DB server 192.168.0.100 FS-2 server 192.168.0.2 *Add two A records in DNS* mydomain.com. 43200 IN A 192.168.0.1 mydomain.com. 43200 IN A 192.168.0.2 I expect, FreeSWITCH instances will mirror each other in this configuration. So, if Bob register on 192.168.0.1 server, and Alice register on 192.168.0.2, they can call each other like local users. Also, in this case, SIP messages from Bob during the call session will sending to any of nodes, depend on DNS resolve. Am I right? Is this correct way to build cluster? Please direct me in right way. BR, Dmitry

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FreeSWITCH library for SRTP/DTLS
by saurabh verrma 09 Apr '16

09 Apr '16

Hi All, I want to use FreeSWITCH library for DTLS/SRTP support. I want to know in freeswitch which library has the support of these features(SRTP/DTLS) ? Is there any application available based on freeswitch library ? Any help would be appreciable. -- *With Warm Regards:* *Saurabh Kumar Verma*

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