Re: [Freeswitch-users] mod_conference performance
Okay, I've performed some additional tests and this is what I've found: * codec**max calls *speex (8kHz) 50 iLBC(8kHz) 50 PCMU(8kHz) 260(approx*) GSM(8kHz)150(approx*) speex(16kHz) 50 G722(16kHz) 90(approx*) * - couldn't trace till the total load of CPU 'cause RDP was timedout due to channel load. Calculated by trend. Still I think Linux tests are necessary. Cheers, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_conference performance (Brian West)
It was one big conference. Robert ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] instant messaging
Hi guys! I'm considering to use SIMPLE protocol for IM in my application, but get a following error trying to send a message from one registered user to another: [ERR] sofia_presence.c:93 Chat proto [sip] from [1...@xx.xxx.xx.xx] to [1...@xx.xxx.xx.xx] 11 Invalid Profile xx.xxx.xx.xx Should presence be enabled in order SIMPLE to work? What additional steps do I have to complete in order to make presence work in FS besides setting "manage_presence" param in SIP profile to true? Best regards, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] instant messaging
Does this message means I've got problems with presence? [ERR] sofia_presence.c:611 DUMP PRESENCE SQL: select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Registered(TLS)','unknown','74.208.167.44',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where (event='presence' or event='dialog') and sub_to_user='1000' and (sub_to_host='xx.xxx.x.xx' or presence_hosts like '%xx.xxx.x.xx%') and (sip_subscriptions.profile_name = 'external' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) EVENT DUMP: Event-Name: [PRESENCE_IN] Core-UUID: [04df3ad6-511b-6d4f-bd4f-517682672b76] ... answer-state: [resubscribe] ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] conference participant from behind NAT
I am a bit confused with what's going on in a following scenario. I have a public FS server with a public conference, that clients are connecting to with my softphone. All of this softphones have STUN option enabled and working, effectively resolving client's public IP address. They also have ICE enabled (but I guess it's not relevant here, since FS doesn't do ICE). Also, media trafic is secured with SRTP. The problem is when one client connects from port-restricted NAT into a conference he can hear sound for some time and he can be heard by other participants, but after awhile sound is gone and neither he hear anything nor he can be heard. Where is the problem? Is it NAT, closing RTP port after some silence period from client? I tried to start conference with waste flag, but without success eventually. The very same person can be contacted through this FS with direct call (being established in proxy_media mode) without any problems, but this is where ICE stuff starts doing its' "magic", I guess. Maybe I should try the same with SRTP disabled? Any help would be apreciated! Best regards, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference participant from behind NAT
Are there ways to escape this timeouts exchanging RTP with FS? Why didn't waste flag help? Maybe I should "flood" channel in both directions? Will CNG on a client side be a good descision? =) ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference participant from behind NAT
в том то все и дело что с тобой мы эту проблему вроде как решили, а у Юры ее никогда не было. и тут на тебе... ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference participant from behind NAT
I am still experiencing problem with lost media in conference on a client behind NAT. This is what I've done - disabled VAD on a NATed client and asked my friend to produce lots of animal sounds in order to keep channel busy. But at the end of minute sounds of wild nature disapeared again. We reproduced that without security with tcp SIP transport and got the same result. Then I started to dig into SIP trace and this is what I found. This client (behind NAT) recieve subsequent INVITE message from FS which seem to destroy dialog and causes client app to close media stream after a session being established normally. I performed the same call from box with public ip and saw no subsequent INVITE's from FS. How come FS sends an INVITE message to already connected client? Is it OK? Should client handle this normally? Below is client's SIP trace: INVITE sip:1.conference...@74.208.167.44:5081;transport=TLS SIP/2.0 ... User-Agent: DoxWox SIP user agent .. SIP/2.0 100 Trying .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. SIP/2.0 407 Proxy Authentication Required .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. ACK sip:1.conference...@74.208.167.44:5081;transport=TLS SIP/2.0 .. SIP/2.0 100 Trying .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. SIP/2.0 183 Session Progress .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. SIP/2.0 200 OK .. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M .. ACK sip:1.conference...@74.208.167.44:5081;transport=tls SIP/2.0 .. *Finally, this message cause media stream closing* INVITE sip:1...@87.184.52.45:64183;transport=tls SIP/2.0 Via: SIP/2.0/TLS 74.208.167.44:5081;branch=z9hG4bK8269NDyXQNjyH Max-Forwards: 70 From: >;tag=vQH234QtN2U8Q To: >;tag=3a231ba86c894ceca81d5021b68d3b6c Call-ID: 37edc38329f64fe98c36cc0a6ddcbd9d CSeq: 121093810 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14953M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 340 v=0 o=FreeSWITCH 1254396503 1254396504 IN IP4 74.208.167.44 s=FreeSWITCH c=IN IP4 74.208.167.44 t=0 0 m=audio 27726 RTP/SAVP 103 101 a=rtpmap:103 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:J6x2UgRVUY8GfbwjCuyyttrtnXnwwuWA9Pt+o3VW ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference participant from behind NAT
And here is a short piece of log from the server side: ... nua(): refersh session after 62 seconds (in [55..65])... send INVITE ... rcv OK... send ACK... rcv BYE... I see now that sdp for natted client has additional lines in OK response compared to client with public ip. Session-Expires: 120;refresher=uas Min-SE: 120 How come that they differs? And how do I resolve this situation? Should client handle these refresher messages normally? Best regards, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference participant from behind NAT
Hi folks! Suddenly I found this http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-February/002015.htmltopic and that explains a lot. >From there I see that sofia sends refresher messages for NATed client in order to check if it still alive. It means I have problems in my client. Sorry for the mess. Cheers, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
[Freeswitch-users] tcp call misses sip message
Hello everyone! I'v got strange problem with incomplete call via tcp transport. When I perform bridged call from one ua (no matter what transport udp or tcp) through FS this call's leg b message sequence (over tcp) lacks finishing SIP message what in it's turn cause the call to be disconnected by the client by timeout. Everything works fine with local calls, so I guess the problem is somewhere between UA and FS. There is no NAT and calls via udp are being established correctly. The problem is with tcp and tls as well. This is the sender's ua SIP trace: TX 1049 bytes Request msg INVITE/cseq=11615 (tdta0486C000) to UDP : RX 348 bytes Response msg 100/INVITE/cseq=11615 (rdata0482806C) from UDP : RX 813 bytes Response msg 407/INVITE/cseq=11615 (rdata0482806C) from UDP : TX 346 bytes Request msg ACK/cseq=11615 (tdta0486EFD0) to UDP : TX 1324 bytes Request msg INVITE/cseq=11616 (tdta0486C000) to UDP : RX 348 bytes Response msg 100/INVITE/cseq=11616 (rdata0482806C) from UDP : RX 1083 bytes Response msg 200/INVITE/cseq=11616 (rdata0482806C) from UDP : TX 360 bytes Request msg ACK/cseq=11616 (tdta04874E38) to UDP : And this is the reciever's SIP trace: RX 1167 bytes Request msg INVITE/cseq=122911315 (rdata04864E10) from tcp : TX 298 bytes Response msg 100/INVITE/cseq=122911315 (tdta0486D010) to tcp : TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp :-- I guess this is where ACK is supposed to arrive Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=1 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=2 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : Retransmiting Response msg 200/INVITE/cseq=122911315 (tdta0486D010), count=0, restart?=3 TX 801 bytes Response msg 200/INVITE/cseq=122911315 (tdta0486D010) to tcp : Sofia profile config: and super-smart dialplan FS 1.0.5pre5 is running on Windows Server 2007SP1 64bit.This issue first occured with 1.0.4 release. Best regards, Robert ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] tcp call misses sip message
but FS does use tcp for that call leg -> RX 1167 bytes ... from *tcp* ...: And after all there can be other SIP transports combinations FS should interconnect... ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] tcp call misses sip message
I've tried to add ;transport=tcp in dialplan bridge application and it has ended up with error on FS with message "can't find registered extension * called_extension*%external_call;transport=tcp" whereas this extension is registered in FS via tcp. Also I tried to reproduce the same scenario with public SIP server and everything worked fine from what I can draw that problem is with my FS configuration or maybe with some FS host's network configuration. Any help will be appretiated. Best regards, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] tcp call misses sip message
Well, I start 2 user agents. Each of them successfully registers as 1000 & 1001 extensions via tcp SIP transport. Then I issue a call, say from 1000 to 1001, and watch it being disconnected in several seconds by recieving client due to abovementioned conditions (no completing answer from FS). Why is it happening??? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] tcp call misses sip message
No, I don't use Xlite. I use my own .Net wrapper around pjsip ua lib. Foreseeing uncertaincies about it's quality I may say that pjsua reference implementation yields the same results in this scenario. Actually I have no doubt that FS is working nicely with tcp and tls as well because I had it working till some moment. And I don't know what the hell happened. =( In order to check if it is something related to my config I switched it back to default and conducted the same test with (urghhh) no luck as well. So now I wonder what could cause this very-very strange behavior? Some issues with network? But why the UDP works then? All traces (FS SIP, FS console, SIP caller and callee's) are here: http://pastebin.com/m2008de4e Regards, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] tcp call misses sip message
Attached is graphical representation of SIP message flow. You can see that for some reason FS doesn't resend to callee an ACK message recieved from caller. Regards, RobertT <>___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] tcp call misses sip message
Yep, I use proxy media. First it started with 1.0.4 release, then I've updated a week or two ago with the latest svn trunk, not sure what was the rev number. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] tcp call misses sip message
OK, this is what I've got. First, I've updated FreeSwitch from trunk to version 15630 and deployed it to my server. Performed a tets and again no magic happened. The link to SIP trace is below. Then I've installed 1.0.4 version to another server (virtual hosting), and performed tha same. And everything went OK. This server's log is below as well. Not working - http://pastebin.com/m2e97985d Working - http://pastebin.com/m3c1e6bfe Also in both cases there is a strange detail - clients' SIP ports are configured to be 5060 and 5061, but what can be seen in trace differs from these values whereas stun resolution shows that there is no NAT (clients connect with ADSL modem). Regards, Robert ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] tcp call misses sip message
You know what, guys? I've just made it working be opening ALL tcp trafic in and out from server by adding two match-all ip filters into local security policy. I can't say I like this solution... Why did this problem appeared with policy matching exact (sofia profiles) ports? Regards, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Requesting testing.
I've a problem building FS rev 15630 on Windows. One of mod_pocketsphinx related projects lack a code file. Regards, Robert. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Presence Change Distribution
You can try to reduce your registration time. I for one made my client apps send PUBLISH message every minute in addition to reduced registration time. Regards, Robert. 2009/12/28 Jerry Richards > Is there a setting to control how fast FS distributes presence changes to > subscribers? Currently, it appears to take several minutes before I see > presence changes. I would like to see them almost instantaneously, if > possible. > > Thanks and Best Regards, > Jerry > > > ___ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
