- FreeSWITCH-users - FreeSWITCH Mailing Lists

SRTP scaling issues
by Matthew Meek 31 Oct '17

31 Oct '17

We have been using Freeswitch as our SBC and core switch for many years without issues. Recently we added public facing softphones using an additional server running Kamailio as TLS and webrtc bridge with FS handling media back to our core FS farm. We run a mix of webrtc and Zoiper softphones. All running SIPS (tls) but webrtc is SRTP (PCMU) and Zoiper is RTP (PCMU). We have been ramping up the webrtc traffic and at some magic point at busy times (~150 concurrent calls) the RTT and Packet loss for the webrtc calls goes crazy (as measured in the browser via RTCP and captured back on our logging servers). RTT jumps from 100ms to 2-3 seconds for most or all webrtc users and packet loss jumps from nothing to troubling on the Sending side back to FS only. Packets received are not an issue on webrtc side... webrtc can hear end caller, but end caller cannot hear webrtc well. Non webrtc traffic (Zoiper) have no reported impact. Server CPU (dual socket 6 core) looks great (no core over 20%) according to nmon... packets per second within reason (10Gb NICs). I am assuming the SRTP has some hidden scaling issue since this is the only difference between the two types of softphones. Has anyone seen this or have a workaround? I am at a loss. I assume if it is encryption overhead the CPU would be saturated and it is not. Why a single direction of SRTP traffic is impacted has me up late at night. A side note... does anyone know of a SIPP replacement that does SRTP so I can test load against this to isolate? Thanks, Matthew Meek

2 2

continue on fail
by Gregor Nanger 31 Oct '17

31 Oct '17

Hi! Need some advice. I want to bidge call to different providers sequentialy and want to specify my own error codes where bridge would considered as failed. For this scenario is variable continue on fail. What bothers me is if I have to set variable before each bridge in dialplan or is suffice on top of dialplan: set continue_on_fail bridge bridge bridge or set continue_on_fail bridge set continue_on_fail bridge set continue_on_fail bridge Thank you, Gregor

1 0

Is there a way I could separate by destination types by hardcoding it in?
by D-Roy Black 31 Oct '17

31 Oct '17

Normally calls coming in from the PSTN to an unprovisioned telephone number are given session:hangup("UNALLOCATED_NUMBER"); Now I have a setup like this, that any call coming in from the PSTN to an unprovisioned telephone number is sent to an IP address, and works fine... session:execute('bridge', 'sofia/internal/gateway/'..dialledNumber..'@100.100.100.100:5061'); But is there a way I could separate by destination types by hardcoding it in? i.e if a caller dialing to 0800 or 0208 UK (sofia/internal/gateway/44800 (sofia/internal/gateway/44208 Any number beginning 44800 should go to session:hangup("UNALLOCATED_NUMBER"); And anything ells would go to the IP address session:execute('bridge', 'sofia/internal/gateway/'..dialledNumber..'@100.100.100.100:5061'); Thanks :) Deltukru

1 0

Prevent mod_callcenter from auto answering channel
by Rovshan Baghirov 31 Oct '17

31 Oct '17

Hello, While caller waits in a queue using mod_callcenter and listens to hold music I do not want caller to be charged by operator. And only start charging when an agent picks up a call. I tried to play some announcement to caller without sending to queue and a caller is able to listen to announcement without being charged. But when I redirect user to mod_callcenter it automatically answers the session. It is coded in mod_callcenter.c line line 2795: /* Make sure we answer the channel before getting the switch_channel_time_table_t answer time */ switch_channel_answer(member_channel); My FreeSWITCH version is: 1.6.19-36-7a77e0b~64bit (-36-7a77e0b 64bit) Now I want to remove this line and add it on line 3051 just before: switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(member_session), SWITCH_LOG_DEBUG, "Member %s <%s> is answered by an agent in queue %s\n", switch_str_nil(switch_channel_get_variable(member_channel, "caller_id_name")), switch_str_nil(switch_channel_get_variable(member_channel, "caller_id_number")), queue_name); I wanted to ask am I doing it right and will it have any side effects except dtmf not working during wait? Here is link to file version I'm talking about: https://freeswitch.org/stash/projects/FS/repos/freeswitch/raw/src/mod/appli… Best Regards, Rovshan

1 0

Questions about sending DTMF outband
by René Weiss 31 Oct '17

31 Oct '17

Hi, I have some questions about sending outband DTMF tones: 1.) What is the right way to send DTMF tones to a parked call? In my tests "uuid_send_dtmf" seems to work only if I first send something (like a silent WAV file) with "uuid_broadcast". (I'm testing with Yealink SIP phones and X-Lite on my computer) Is this the right way to do it, or should I do something else to "initialize" the connection? 2.) Is there a way to detect if all DTMF tones have been played? It seems to me that "uuid_send_dtmf" returns before all tones have been played and I don't see any Freeswitch event triggered after the DTMF sequence has ended. 3.) Can I make "uuid_send_dtmf" play faster? "uuid_send_dtmf <uuid> 1234@1" plays much slower than for example "uuid_broadcast <uuid> tone_stream://w=100;d=100;1234" probably because of the "Queue digit delay of 40m" after each tone. Can I reduce this delay or is this a limitation of the DTMF outband protocol? Thank you in advance, René

1 0

Optimal trunk capacity filling algorithm
by Adam Raszynski 30 Oct '17

30 Oct '17

Hi All, I'm looking for a solution to a problem that is not FreeSWITCH specific, but for sure some of you have dealt with this, and maybe someone can suggest optimal solution, so I could not re-invent the wheel It's all about proper filling of available termination trunk capacity, but not the trivial one with simple LCR Let's say we have two trunks: - Trunk 1 - capacity 100 channels, termination cost 1 cent - Trunk 2 - capacity 200 channels, termination cost 3 cent And we have clients sending wholesale traffic with random number of channels: - Client 1 - sell rate 2 cent - Client 2 - sell rate 4 cent As you see we call only sell to client 1 with profit using trunk 1. If we use LCR and client 2 has more traffic, than it will fill trunk 1 capacity and client 2 will hit trunk 2 and we will start to loose Sticking client 1 to trunk 1 and client 2 to trunk 2 is also not optimal. Number of channels is more or less random. If we stick client 2 to trunk 2 when client 1 is not using all 100 channels, than we loose additional 1 cent of margin I think about some form of capacity-aware dynamic routing - send client 2 traffic to trunk 1 only if client 1 cannot fill trunk 1, maybe using some capacity averages from last X minutes. Any suggestions how to approach this problem? Kind Regards

2 1

How to encrypt user extension password
by Shisheer Teli 30 Oct '17

30 Oct '17

Dear Team, I am able to see user password in user extension configuration. Can anyone help to encrypt user extension password <include> <user id="1111"> <params> <param name="password" value="abcd123"/> <param name="vm-password" value="1111"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1111"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_number" value="1111"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="support"/> </variables> </user> </include> -- Regards, Shisheer T *Save paper, save trees.... Please do not print this e-mail unless it is absolutely necessary.*

2 1

FS_PATH for Gateways
by Sharath Kumar 27 Oct '17

27 Oct '17

Can we add an fs_path when dialing out via a gateway ? I know this has been asked before and in the archives it has been said to support and also not support fs_path in a gateway. Just wanted to confirm. thank you, Shaks.

5 8

Multiple "Changing expire time to 60 by request of proxy" requests result in delayed audio
by Erik M. Devane - Comms Guy 27 Oct '17

27 Oct '17

Hello, I have a fairly generic upstream VoIP provider. Every day or so of calls, I receive: 2017-10-26 16:54:06.100048 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:54:31.120584 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:55:09.139687 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:55:48.160453 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:56:04.180054 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:56:26.180559 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:56:43.199818 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:57:24.219901 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:58:08.260016 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:58:29.259860 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 2017-10-26 16:59:00.281354 [DEBUG] sofia_reg.c:2402 Changing expire time to 60 by request of proxy sip:69.71.29.31 If there are just one or two, nothing seems to happen. When I see ten or more, there's an audio delay of five seconds or so. Can anyone help point me in the right direction? Thank you, Erik

2 1

Gigaset auto answer
by Nicola von Thadden 27 Oct '17

27 Oct '17

Hi, I'm trying to get auto answer working on my Gigaset N510 IP DECT system. I made the changes to the DECT config (enabled "Call Manager" for the extension I want to use_ and tried to call it using the default prefix (8xxxx). Unlike the other phones I work with, the Gigaset device requires authentication for that. Since it is a subscriber to the freeswitch system, the system does not have any information to fulfill it: [ERR] sofia_reg.c:2561 Cannot locate any authentication credentials to complete an authentication request for realm '"test"' The INVITE itself looks as I would expect it to look but the phone answers with: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 46.38.244.168;rport=5060;branch=z9hG4bKK94yXK0rDUcXj From: "Nico" <sip:1000@xxxxx>;tag=6gtHSvy9p62HH To: <sip:1012@xxxxx:5060>;tag=2471884501 Call-ID: dee1f01e-2fc3-1236-8f91-52542680391a CSeq: 113890892 INVITE Contact: <sip:1012@xxxxx:5060> WWW-Authenticate: Digest realm="test", nonce="d0533fa35f7a53f86112305a818d732", algorithm=MD5, qop="auth" User-Agent: N510 IP PRO/42.243.00.000.000 Content-Length: 0 Does anyone have an idea how I can either get freeswitch to retry with authentication or how to get the Gigaset device to accept the call without further auth? Thanks Nico

2 4