- sr-users - kamailio.org

Kamailio World thanks
by Daniel Pocock 16 May '19

16 May '19

Hi Daniel and Kamailio team, Well done on another successful Kamailio World in 2019. As usual, it was a great opportunity to meet people and learn new things. Regards, Daniel

2 1

siputils add_uri_param not available in KEMI python3
by Patrick Murphy 15 May '19

15 May '19

Hello, I am using the following version of Kamailio and using app_python3 with KEMI but I cannot find any exported function of add_uri_param from the siputils module. Any idea how I can perform: KSR.siputils.add_uri_param("name=value") ? version: kamailio 5.2.2 (x86_64/linux) 9260a2 Cheers, Pat.

1 1

Using a single virtual domain in a multi-domain setup
by Rhys Hanrahan 15 May '19

15 May '19

Hi All, I am trying to setup Kamailio to allow me to have multiple different domains (and IPs) treated as one. I am using the domain module for this. As an example I have: * A handset registered to Kamailio by IP address. * A SIP trunk between Asterisk and Kamilio using an FQDN that uses SRV records for redundancy on Asterisk's side. I have added all domains and IPs under the same "domain identifier". It seems the "old" domain module now uid_domain had this capability of a single virtual domain, but is this possible with the more current domain.so module? From what I've said the uid_* modules older modules from the SIP Router project, and should be replaced by more recent Kamailio modules? - https://www.kamailio.org/docs/modules/5.0.x/modules/uid_domain.html VS. - https://kamailio.org/docs/modules/5.1.x/modules/domain.html MariaDB [kamailio]> select * from domain; +----+------------------------------------+------------+---------------------+ | id | domain | did | last_modified | +----+------------------------------------+------------+---------------------+ | 1 | 12.34.56.53 | vocalpoint | 2000-01-01 00:00:01 | | 2 | 23.34.56.54 | vocalpoint | 2000-01-01 00:00:01 | | 3 | sbc.foo | vocalpoint | 2000-01-01 00:00:01 | | 4 | sbc.nsw.foo | vocalpoint | 2000-01-01 00:00:01 | | 5 | sbc.vic.foo | vocalpoint | 2000-01-01 00:00:01 | | 6 | sbc.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 7 | sbc.nsw.foo2 | vocalpoint | 2000-01-01 00:00:01 | | 8 | sbc.vic.foo2 | vocalpoint | 2000-01-01 00:00:01 | +----+------------------------------------+------------+---------------------+ 8 rows in set (0.000 sec) I managed to get registrations working by hardcoding a static realm in the config, but when when usrloc does a lookup on a handset, Kamailio can't find user(a)sbc.nsw.foo and Kamailio returns 404. It looks like this is because in the locations table it is listed with a domain of 12.34.56.53 (because the handset is registered by IP). How should this work normally? Or is it even possible? Thanks, Rhys.

2 4

Sip proxy for Asterisk with RFC3966 and 100Rel
by paolo.visnoviz＠vipcomputers.it 15 May '19

15 May '19

Dear Sirs, a telco provider give us a trunk that require 100rel and TEL RFC 3966. Usually we use Asterisk. Now Asterisk 16 with PJSIP support 100Rel, but not RFC3966, and they don't actually works to implement it. Asterisk chan_sip, instead, support RFC3966, but not 100Rel and is not actively developed on by Digium or Sangoma. So I'm stuck. Now, do you think it was possible to use Kamailio to act like a proxy sip but implementing RFC3966 for incoming and outgoing calls and RFC3966 and 100Rel for outgoing calls? And it is possible to connect all to an Asterisk server? Thank you in advance Best regards. Paolo Visnoviz

3 9

Possible memory leak in siptrace module in kamailio 5.1/5.2
by davy van de moere 15 May '19

15 May '19

Dear, I'm using siptrace within a python handler in the KEMI framework. KSR.siptrace.sip_trace_dst("sip:<myhomerip>:9060") I notice an ever increasing usage of memory, until the system runs out of memory. (definition of a memory leak :) ). Looking at the memory information kamailio is giving me I notice this: May 14 05:49:48 server kamailio[36452]: NOTICE: PYT {INVITE} {1769595029-1513210001-1955500877} {1} : qm_sums: count= 79286 size= 5812048 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) May 14 05:51:17 server kamailio[36452]: NOTICE: PYT {REGISTER} { 4_1419955802(a)192.168.0.153} {2582} : qm_sums: count= 79335 size= 5814840 bytes from siptrace: siptrace.c: ki_sip_trace_dst_cid(729) I notice this on kamailio 5.1.8 (from a debian repo), and on kamailio 5.2.2 (from source). I can reproduce the problem very easily, and I can apply patches in the source to try out things. Could I be wrongly using siptrace? Can I give you more debugging info? Kind regards, Davy

3 4

define a DBURL with SLQOPS fails
by David Villasmil 14 May '19

14 May '19

Hello guys, So, i'm defining: DBURL "mysql://user:password@dbhost/dbname" and later on... modparam("sqlops","sqlcon","cb=>"DBURL) and this fails miserably.. What do do on this case? Thanks all! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 12

Npdi question
by Duarte Rocha 14 May '19

14 May '19

Greetings, Regarding portability issues, is there a difference between "npdi" and "npdi=yes" ? Are there any more possible values besides "yes" and no value? Best Regards, Duarte Rocha

1 0

sdp_remove_codecs_by_id does not remove 101
by Juha Heinanen 14 May '19

14 May '19

I tried to use sdp_remove_codecs_by_id("101") to remove "codec" 101 (dtmf) from sdp, but nothing happened. It is still there: m=audio 31128 UDP/TLS/RTP/SAVPF 101 96. c=IN IP4 192.168.43.88 a=label:1. a=rtcp-rsize. a=ssrc:2880178656 cname:sip:test@test.tutpro.com. a=rtpmap:101 telephone-event/8000. a=rtpmap:96 opus/48000/2. a=fmtp:101 0-15. Is that by design or a bug? If not a bug, is there some other means to remove 101? -- Juha

2 2

sip trunk with kamailio
by arish haque 14 May '19

14 May '19

Hi, I am trying the SIP trunk from kamailio with following configurations - cfg_get.pstn.gw_ip="10.0.x.y route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(!($rU=~"^(\+|00)918884978235")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } I am able to see kamailio sending INVITE to SIP Provider but no response back. Any other, configuration is required which I'm missing. I have attached the tcpdump snapshot of the above scenario. Thanks, Arish

2 1

Theoritical Anycast Setup
by Daniel Oakes 14 May '19

14 May '19

Hi there, Long time user, 1st time poster - apologies if someone has posted this previously. We have a product from a vendor that utilizes Kamailio as a SIP router - but we're wanting to extend it a little bit by the use of anycast (we're a small ISP). Basically the scenario is that we want to utilize Quagga via BGP to anycast an address, which addresses a glaring load balancing issue with have with people who use only A records. Problem is anycast doesn't work so well in this scenario, just by adding the Anycast as a listen address on the proxy. What we'd want to do is rewrite the SIP so that all responses come back out saying to use the 'real ip'. So I have two variables - say REAL_IP_ADDRESS and ANYCAST_IP_ADDRESS - request will come in via ANYCAST, but I want all responses from then on to be via REAL_IP_ADDRESS. We have six Kamailio boxes in various geographical locations, so the idea is they hit one based on closest, and we rewrite it so they end up with the real ip address to talk to. Anyone done something similar? I'm guessing possibly, but I can't find anything specific enough for me to use. Help appreciated! Regards, Daniel

2 1

No audio after IP address change
by Chris Scheid 11 May '19

11 May '19

Hi, we are using rtpengine with Kamailio using DTLS-SRTP and SRTP making audio calls from two Android devices. Regular audio calls work fine but we have issues with the following scenario: 1. Device 1 (72.66.83.8, on WiFI) calls Device 2 (174.204.14.29). SIP server has IP 209.182.216.71. 2. Device 2 switches networks from WiFI to 3G (new IP: 97.47.70.223). Re-invite flow completes. 3. Any audio from Device 2 cannot be heard on Device 1. But audio from the Device1 can be heard on the Device 2. In the rtpengine stats I see that there were some packets sent from the SIP server to the Device 1 after network change: [1557513316.000308] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: ------ Media #1 (audio over UDP/TLS/RTP/SAVP) using iLBC/8000 [1557513316.000318] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20052 <> 97.47.70.223:4667 , SSRC 77c38349, 55 p, 3410 b, 0 e, 32 ts [1557513316.000325] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20053 <> 97.47.70.223:4640 (RTCP), SSRC 77c38349, 12 p, 1120 b, However, in my UDP trace, I don't see any communication between the SIP server and the Device 1 after network change (IP 97.47.70.223). This seems to be the reason why we don't get audio on Device 1. I'm looking for direction on how to troubleshoot this issue. I'm not so familiar with rtpengine. Any assistance greatly appreciated. Thanks! RTP Engine Config: [rtpengine] #####table = 0 listen-ng = 127.0.0.1:5050 timeout = 60 silent-timeout = 3600 tos = 184 port-min = 20000 port-max = 60000 interface = 209.182.216.71 log-level = 7 log-stderr = false log-facility = local0 START CALL --------------------------------------------------------------------------------------------------------------------- [1557513231.973294] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'offer' from 127.0.0.1:45431 [1557513231.973357] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'offer' from 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502030 3766502030 IN IP4 192.168.1.200 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5095 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 192.168.1.200 b=TIAS:64000 a=rtcp:5096 IN IP4 192.168.1.200 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2009301833 cname:32ababda5d2c847b a=setup:actpass a=fingerprint:SHA-256 4F:69:B6:89:51:98:6C:A4:63:CF:F7:CD:22:D2:BE:FD:6E:3F:F4:2A:8B:39:AD:E4:C0:8D:AE:BD:A3:70:12:49 ", "ICE": "remove", "DTLS": "passive", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9 799-0557b76cc340", "received-from": [ "IP4", "72.66.83.8" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "command": "offer" } [1557513231.973444] NOTICE: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating new call [1557513231.973496] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Default sink codec is speex/16000 [1557513231.973503] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for iLBC/8000 [1557513231.973508] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for speex/16000 [1557513231.973512] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/16000 [1557513231.973515] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for speex/8000 [1557513231.973519] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/8000 [1557513231.973523] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for speex/32000 [1557513231.973526] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/32000 [1557513231.973530] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for GSM/8000 [1557513231.973534] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec GSM/8000 [1557513231.973538] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for PCMU/8000 [1557513231.973542] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMU/8000 [1557513231.973545] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for PCMA/8000 [1557513231.973549] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMA/8000 [1557513231.973545] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for PCMA/8000 [1557513231.973549] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMA/8000 [1557513231.973552] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for G722/8000 [1557513231.973556] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec G722/8000 [1557513231.973560] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for telephone-event/8000 [1557513231.974291] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 192.168.1.200:5095 [1557513231.974561] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 192.168.1.200:5096 [1557513231.974874] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: offer time = 0.001488 sec [1557513231.974882] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'offer' from 127.0.0.1:45431 [1557513231.974900] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'offer' to 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502030 3766502030 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20032 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2009301833 cname:32ababda5d2c847b a=rtpmap:104 iLBC/8000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20033 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:G9MvyRPBAVFsrMypvr1EAunZiX/vc04sXEq4mXvl a=setup:actpass a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513232.077386] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Processing incoming DTLS packet [1557513232.078219] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Sending DTLS packet [1557513232.189220] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Processing incoming DTLS packet [1557513232.189868] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: DTLS handshake successful [1557513232.189899] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: DTLS-SRTP successfully negotiated [1557513232.189905] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: SRTP keys, incoming: [1557513232.189914] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: --- AES_CM_128_HMAC_SHA1_80 key q67B3h3GP/dhbQeKsgBnvA== salt EcoNJ6Hqj0lRtvZ8Z+Y= [1557513232.189919] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: SRTP keys, outgoing: [1557513232.189923] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: --- AES_CM_128_HMAC_SHA1_80 key 0nTo2gy4XJ2GXlroyyH6nA== salt fbIfUIa4IPiYLvn5CkI= [1557513232.189929] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Sending DTLS packet confirmed [1557513243.648460] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SRTCP output wanted, but no crypto suite was negotiated [1557513243.648486] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513243.648496] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 9fda818d: FL 205 TL 6486816 HSR 613804890 J 3100337071 LSR 1419186059 DLSR 531075031 [1557513243.648505] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513243.649142] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513243.649158] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about d4b1278d: FL 23 TL 2271802 HSR 278642673 J 2408385537 LSR 3 653820601 DLSR 1701759944 [1557513243.649169] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513243.777813] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'answer' from 127.0.0.1:47861 [1557513243.777860] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'answer' from 127.0.0.1:47861: { "sdp": "v=0 o=- 3766502031 3766502032 IN IP4 100.81.62.76 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5079 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 100.81.62.76 b=TIAS:64000 a=rtcp:5080 IN IP4 100.81.62.76 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2006816651 cname:1c0965113e9b1046 a=setup:active a=fingerprint:SHA-256 AD:2D:0F:A6:11:DD:75:CC:3C:50:3A:9A:F0:29:20:8D:95:BC:3E:2A:A0:34:F6:67:14:A7:87:EB:8D:66:77:B1 ", "ICE": "remove", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9799-0557b76cc340", "received-from": [ "IP4", "174.204.14.29" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "to-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "command": "a nswer" } [1557513243.777998] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for iLBC/8000 [1557513243.778010] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Creating codec handler for telephone-event/8000 [1557513243.778088] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 100.81.62.76:5079 [1557513243.778117] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: DTLS: Peer certificate accepted [1557513243.778123] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 100.81.62.76:5080 [1557513243.778166] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: answer time = 0.000267 sec [1557513243.778175] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'answer' from 127.0.0.1:47861 [1557513243.778187] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'answer' to 127.0.0.1:47861: { "sdp": "v=0 o=- 3766502031 3766502032 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20052 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2006816651 cname:1c0965113e9b1046 a=rtpmap:104 iLBC/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20053 a=setup:passive a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513243.864786] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513243.865568] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513243.912858] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513243.913111] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS: Peer certificate accepted [1557513243.913627] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS handshake successful [1557513243.913668] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS-SRTP successfully negotiated [1557513243.913675] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, incoming: [1557513243.913682] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key +C2ck1NPvsfHYoKflimqaA== salt xM4aFoL60/ePpmQJsT8= [1557513243.913686] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, outgoing: [1557513243.913689] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key VvTNUyescFh+vNj16/fePw== salt m3EuTJjAesJHaSXfqKU= [1557513243.913694] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513244.475362] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513244.475405] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 3873048209 PC 2516599107 OC 673505029 NTP TS 1161195675/4 008032614=1161195675.933193 [1557513244.475412] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about bb94938e: FL 243 TL 11665253 HSR 1302964504 J 2830082235 LS R 1964993324 DLSR 1625750139 [1557513244.475422] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513244.492749] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SRTCP output wanted, but no crypto suite was negotiated [1557513244.492775] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513244.492791] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2981125880 PC 4052113656 OC 2653921327 NTP TS 4076801750/ 1160081551=4076801750.270103 [1557513244.492797] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 3813d27e: FL 75 TL 11393233 HSR 1719559869 J 944495881 LSR 790240102 DLSR 447292567 [1557513244.492804] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513247.217290] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20032]: Confirmed peer address as 174.204.14.29:10523 [1557513248.958592] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513248.958638] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 3148803183 PC 137367242 OC 1506652935 NTP TS 3606800652/1 621372750=3606800652.377505 [1557513248.958647] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 468fa6ae: FL 57 TL 15596558 HSR 3764947872 J 3769875796 LSR 4259175195 DLSR 3903464346 [1557513248.958657] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513248.958667] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Confirmed peer address as 174.204.14.29:10496 i[1557513249.259063] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513249.259108] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 3424021083 PC 3823893658 OC 584056375 NTP TS 3244065141/3 391763074=3244065141.789706 [1557513249.259117] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about e7313af6: FL 98 TL 10263186 HSR 2762600672 J 838876573 LSR 3367219470 DLSR 50549698 [1557513249.259131] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513249.259142] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Confirmed peer address as 72.66.83.8:5096 t[1557513249.562239] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Confirmed peer address as 72.66.83.8:5095 SWITCH NETWORKS [1557513253.843937] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513253.844019] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 1364757765 PC 1063357287 OC 2194907961 NTP TS 2472417766/ 1884000851=2472417766.438653 [1557513253.844029] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 77b8933: FL 163 TL 2305413 HSR 2377964402 J 118139039 LSR 2 460485565 DLSR 3416921415 [1557513253.844040] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513254.590466] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513254.590508] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2527484095 PC 1878948628 OC 3882176963 NTP TS 213338522/6 38146641=213338522.148580 [1557513254.590517] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about b7800717: FL 93 TL 16503124 HSR 338254320 J 1043490285 LSR 801212789 DLSR 2849120558 [1557513254.590530] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513258.679347] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SRTCP output wanted, but no crypto suite was negotiated [1557513258.679385] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513258.679411] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 1747525463 PC 239751946 OC 2859161225 NTP TS 3835334254/3 577221347=3835334254.832887 [1557513258.679418] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about e03bd142: FL 40 TL 11515285 HSR 3870936491 J 4150119236 LSR 2903910302 DLSR 700479064 [1557513258.679427] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513259.940740] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SRTCP output wanted, but no crypto suite was negotiated [1557513259.940770] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513259.940793] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2745657868 PC 580822102 OC 4072320279 NTP TS 252333929/20 93935903=252333929.487532 [1557513259.940801] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 476ec510: FL 61 TL 5399464 HSR 3682903462 J 1600791504 LSR 497094189 DLSR 826539320 [1557513259.940814] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513263.532695] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 200 [1557513263.532722] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: SR from 779d978b: RTP TS 301750475 PC 2434799802 OC 863475295 NTP TS 3625503376/30 14095725=3625503376.701774 [1557513263.532729] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about 228989b7: FL 148 TL 10638176 HSR 2137788972 J 2678688015 LS R 2118438561 DLSR 1024262683 [1557513263.532738] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513264.051305] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513264.051352] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1190725747 PC 3072491822 OC 537479863 NTP TS 817717026/42 82321058=817717026.997056 [1557513264.051359] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 920db8ff: FL 98 TL 6598240 HSR 2045142812 J 3171281598 LSR 2224002690 DLSR 2514604823 [1557513264.051369] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513264.058891] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 201 [1557513264.058920] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 5c8f2711: FL 24 TL 1700303 HSR 2993289918 J 618517078 LSR 1 835926259 DLSR 3106454771 [1557513264.058891] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 201 [1557513264.058920] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 5c8f2711: FL 24 TL 1700303 HSR 2993289918 J 618517078 LSR 1 835926259 DLSR 3106454771 [1557513264.058934] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513264.266612] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'offer' from 127.0.0.1:60736 [1557513264.266654] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'offer' from 127.0.0.1:60736: { "sdp": "v=0 o=- 3766502030 3766502031 IN IP4 192.168.1.200 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5097 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 10.255.81.66 b=TIAS:64000 a=rtcp:5098 IN IP4 10.255.81.66 a=sendrecv a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2009301833 cname:32ababda5d2c847b a=setup:actpass a=fingerprint:SHA-256 4F:69:B6:89:51:98:6C:A4:63:CF:F7:CD:22:D2:BE:FD:6E:3F:F4:2A:8B:39:AD:E4:C0:8D:AE:BD:A3:70:12:49 ", "ICE": "remove", "DTLS": "passive", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9 799-0557b76cc340", "received-from": [ "IP4", "97.47.70.223" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "to-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d 00", "command": "offer" } [1557513264.266814] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Default sink codec is speex/16000 [1557513264.266826] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/16000 [1557513264.266831] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/8000 [1557513264.266835] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec speex/32000 [1557513264.266838] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec GSM/8000 [1557513264.266842] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMU/8000 [1557513264.266846] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec PCMA/8000 [1557513264.266850] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sink supports codec G722/8000 [1557513264.267614] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet [1557513264.268089] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet [1557513264.268146] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet [1557513264.268220] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 10.255.81.66:5097 [1557513264.268627] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: set FILLED flag for stream 10.255.81.66:5098 [1557513264.269072] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: offer time = 0.002384 sec [1557513264.269084] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'offer' from 127.0.0.1:60736 [1557513264.269098] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'offer' to 127.0.0.1:60736: { "sdp": "v=0 o=- 3766502030 3766502031 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20064 UDP/TLS/RTP/SAVP 104 98 97 99 3 0 8 9 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2009301833 cname:32ababda5d2c847b a=rtpmap:104 iLBC/8000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:99 speex/32000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20065 a=setup:actpass a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513264.355109] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SRTCP output wanted, but no crypto suite was negotiated [1557513264.355140] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 201 [1557513264.355150] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 511a36de: FL 144 TL 3729734 HSR 2556650506 J 4034165594 LSR 900570716 DLSR 3118967760 [1557513264.355166] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513264.355186] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513264.355193] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about aea81208: FL 180 TL 10378624 HSR 1586287909 J 2246375086 LS R 3231023203 DLSR 4073439206 [1557513264.355201] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513264.355210] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Switching local interface to 209.182.216.71:20033 [1557513264.358206] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: Calling handler for RTCP packet type 201 [1557513264.358229] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: RR from 779d978b about b59d47bc: FL 242 TL 13886413 HSR 3966401064 J 2679493971 LS R 3040876731 DLSR 2668321392 [1557513264.358239] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20033]: No payload type known for RTCP RR, discarding [1557513264.358262] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20064]: SRTP output wanted, but no crypto suite was negotiated [1557513264.425636] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'answer' from 127.0.0.1:45431 [1557513264.425679] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'answer' from 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502031 3766502033 IN IP4 100.81.62.76 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 5079 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 100.81.62.76 b=TIAS:64000 a=rtcp:5080 IN IP4 100.81.62.76 a=sendrecv a=rtpmap:104 iLBC/8000 a=rtpmap:104 iLBC/8000 a=fmtp:104 mode=30 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ssrc:2006816651 cname:1c0965113e9b1046 a=setup:active a=fingerprint:SHA-256 AD:2D:0F:A6:11:DD:75:CC:3C:50:3A:9A:F0:29:20:8D:95:BC:3E:2A:A0:34:F6:67:14:A7:87:EB:8D:66:77:B1 ", "ICE": "remove", "replace": [ "origin", "session-connection" ], "transport-protocol": "UDP/TLS/RTP/SAVP", "call-id": "05c7dcfe-06f3-4630-9799-0557b76cc340", "received-from": [ "IP4", "174.204.14.29" ], "from-tag": "2684eff8-3602-41ed-aca2-9f7938362462", "to-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "command": "a nswer" } [1557513264.426806] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: answer time = 0.001099 sec [1557513264.426820] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'answer' from 127.0.0.1:45431 [1557513264.426831] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'answer' to 127.0.0.1:45431: { "sdp": "v=0 o=- 3766502031 3766502033 IN IP4 209.182.216.71 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 20052 UDP/TLS/RTP/SAVP 104 96 c=IN IP4 209.182.216.71 b=TIAS:64000 a=ssrc:2006816651 cname:1c0965113e9b1046 a=rtpmap:104 iLBC/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:104 mode=30 a=fmtp:96 0-16 a=sendrecv a=rtcp:20053 a=setup:passive a=fingerprint:sha-1 EE:4A:ED:7D:5A:94:A2:FF:F2:09:64:07:41:FC:15:80:D3:F9:C0:B4 ", "result": "ok" } [1557513264.700442] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513264.701235] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513264.892393] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Processing incoming DTLS packet [1557513264.893153] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS handshake successful [1557513264.893193] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: DTLS-SRTP successfully negotiated [1557513264.893202] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, incoming: [1557513264.893211] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key dob4sywpHJJ8qNgFzemlzQ== salt y7/1RgzeAtPkG+GfMLI= [1557513264.893216] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: SRTP keys, outgoing: [1557513264.893222] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: --- AES_CM_128_HMAC_SHA1_80 key ontM1nC56XojOkmzb8520g== salt Hom6TrHovUBUaF1g6Iw= [1557513264.893228] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Sending DTLS packet [1557513269.223923] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20052]: Confirmed peer address as 97.47.70.223:4667 [1557513269.337491] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513269.337543] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 402044265 PC 1976845075 OC 2962195809 NTP TS 1329245270/2 977782988=1329245270.693319 [1557513269.337552] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about c1e9445b: FL 196 TL 15740765 HSR 1231260891 J 3373413243 LS R 2989889476 DLSR 1721296716 [1557513269.337562] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513269.337571] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Confirmed peer address as 97.47.70.223:4640 [1557513269.355712] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513269.355744] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 46888194 PC 1178262997 OC 1943197043 NTP TS 748304878/374 [1557513269.355749] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 4d78530a: FL 59 TL 15587360 HSR 641166863 J 455122196 LSR 2 981586141 DLSR 1538485973 [1557513269.355755] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513269.355763] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Confirmed peer address as 174.204.14.29:10496 [1557513269.355768] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Switching local interface to 209.182.216.71:20065 [1557513269.376119] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 201 [1557513269.376149] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 60b442ee: FL 185 TL 2381849 HSR 1868471658 J 3454737373 LSR 3540730944 DLSR 753951698 [1557513269.376164] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513269.967467] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20064]: Confirmed peer address as 174.204.14.29:10523 s[1557513274.064040] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SRTCP output wanted, but no crypto suite was negotiated [1557513274.064063] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513274.064084] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 2043506601 PC 348629863 OC 1509649344 NTP TS 4002640369/1 330623176=4002640369.309810 [1557513274.064090] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 1b7ce7ec: FL 17 TL 10773844 HSR 2006802227 J 3496394774 LSR 1855863287 DLSR 207902258 [1557513274.064105] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 1b7ce7ec) - deleting SSRC 9fda8 18d [1557513274.064114] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding w[1557513274.337285] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513274.337341] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 3588538011 PC 4113250253 OC 3762757735 NTP TS 3675530209/ 2088363704=3675530209.486235 [1557513274.337352] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 1904c0d9: FL 190 TL 1277330 HSR 3700606795 J 2076634040 LSR 2826708796 DLSR 3084916091 [1557513274.337367] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add 1904c0d9) - deleting SSRC d4b12 78d [1557513274.337378] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding itch complete[1557513278.796840] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513278.796890] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1179546147 PC 253436483 OC 401514196 NTP TS 591585900/307 4094117=591585900.715743 [1557513278.796900] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about f25fab39: FL 196 TL 1861219 HSR 1237831119 J 2726348211 LSR 3841137799 DLSR 843496811 [1557513278.796913] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add f25fab39) - deleting SSRC bb949 38e [1557513278.796921] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513279.310594] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SRTCP output wanted, but no crypto suite was negotiated [1557513279.310626] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513279.310646] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 624630617 PC 1167418798 OC 2870042862 NTP TS 2807429790/9 64181525=2807429790.224491 [1557513279.310652] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about bb84140b: FL 193 TL 5406510 HSR 1470777656 J 207786249 LSR 3684806947 DLSR 307709608 [1557513279.310669] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add bb84140b) - deleting SSRC 3813d 27e [1557513279.310679] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513279.678189] ERR: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20064]: SRTP output wanted, but no crypto suite was negotiated [1557513283.537471] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513283.537514] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1085702245 PC 129819128 OC 1350967259 NTP TS 967851883/21 63508869=967851883.503731 [1557513283.537521] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 169e1507: FL 46 TL 3777017 HSR 3092777669 J 3495001522 LSR 3630093505 DLSR 120738828 [1557513283.537537] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 169e1507) - deleting SSRC 468fa 6ae [1557513283.537546] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513284.300047] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513284.300094] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 3660274611 PC 3339904092 OC 976632286 NTP TS 3250964209/2 354170892=3250964209.548123 [1557513284.300103] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about a9cab1ac: FL 9 TL 7152183 HSR 1071025251 J 612018816 LSR 12 43749409 DLSR 721434160 [1557513284.300119] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add a9cab1ac) - deleting SSRC e7313 af6 [1557513284.300130] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding bye [1557513286.941239] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Received command 'delete' from 127.0.0.1:49479 [1557513286.941278] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Dump for 'delete' from 127.0.0.1:49479: { "call-id": "05c7dcfe-06f3-4630-9799-0557b76cc340", "received-from": [ "IP4", "174.204.14.29" ], "from-tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "command": "delete" } [1557513286.941434] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Scheduling deletion of call branch 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00' (via-branch '') in 3 0 seconds [1557513286.941444] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: delete time = 0.000152 sec [1557513286.941474] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Replying to 'delete' from 127.0.0.1:49479 [1557513286.941618] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Response dump for 'delete' to 127.0.0.1:49479: { "created": 1557513231, "created_us": 973175, "last signal": 1557513264, "SSRC": { "3046983612": { }, "2450372863": { }, "1622426350": { }, "2930250248": { }, "419741913": { }, "579439031": { }, "136 0672478": { }, "3145995275": { }, "125536563": { }, "2848633260": { }, "3762016578": { }, "379458823": { }, "1552885521": { }, "3078620951": { }, "12997 31210": { }, "461170668": { }, "2009301833": { }, "3253290075": { }, "1198441744": { }, "4066356025": { }, "2006816651": { } }, "tags": { "c71ac9de-9bf5- 4f39-9fda-6bd3aaf97d00": { "tag": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "created": 1557513231, "in dialogue with": "2684eff8-3602-41ed-aca2-9f7938362462", "me dias": [ { "index": 1, "type": "audio", "protocol": "UDP/TLS/RTP/SAVP", "streams": [ { "local port": 20064, "endpoint": { "family": "IPv4", "address": "174.204. 14.29", "port": 10523 }, "advertised endpoint": { "family": "IPv4", "address": "100.81.62.76", "port": 5079 }, "crypto suite": "AES_CM_128_HMAC_SHA1_80", "last packet": 1557513286, "flags": [ "RTP", "filled", "confirmed", "kernelized", "no kernel support", "DTLS fingerprint verified" ], "SSRC": 2006816651, "stats": { " packets": 595, "bytes": 40090, "errors": 0 } }, { "local port": 20065, "endpoint": { "family": "IPv4", "address": "174.204.14.29", "port": 10496 }, "advertised endpoint": { "family": "IPv4", "address": "100.81.62.76", "port": 5080 }, "crypto suite": "AES_CM_128_HMAC_SHA1_80", "last packet": 1557513284, "flags": [ "RTCP ", "filled", "confirmed", "kernelized", "no kernel support" ], "SSRC": 2006816651, "stats": { "packets": 15, "bytes": 1306, "errors": 0 } } ], "flags": [ "initi alized", "send", "recv", "DTLS-SRTP", "DTLS role passive", "ICE controlling" ] } ] }, "2684eff8-3602-41ed-aca2-9f7938362462": { "tag": "2684eff8-3602-41ed-aca2- 9f7938362462", "created": 1557513231, "in dialogue with": "c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00", "medias": [ { "index": 1, "type": "audio", "protocol": "UDP/TL S/RTP/SAVP", "streams": [ { "local port": 20052, "endpoint": { "family": "IPv4", "address": "97.47.70.223", "port": 4667 }, "advertised endpoint": { "family": " IPv4", "address": "10.255.81.66", "port": 5097 }, "crypto suite": "AES_CM_128_HMAC_SHA1_80", "last packet": 1557513284, "flags": [ "RTP", "filled", "confirmed", "kernelized", "no kernel support", "DTLS fingerprint verified" ], "SSRC": 2009301833, "stats": { "packets": 55, "bytes": 3410, "errors": 0 } }, { "local port": 20053, "endpoint": { "family": "IPv4", "address": "97.47.70.223", "port": 4640 }, "advertised endpoint": { "family": "IPv4", "address": "10.255.81.66", "port": 5098 }, "last packet": 1557513283, "flags": [ "RTCP", "filled", "confirmed", "kernelized", "no kernel support" ], "SSRC": 2009301833, "stats": { "packets": 10, "bytes": 936, "errors": 0 } } ], "flags": [ "initialized", "send", "recv", "DTLS-SRTP", "DTLS role passive" ] } ] } }, "totals": { "RTP": { "packets": 650, "bytes": 43500, "errors": 0 }, "RTCP": { "packets": 25, "bytes": 2242, "errors": 0 } }, "result": "ok" } [1557513287.087244] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513287.087244] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 200 [1557513287.087282] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SR from 77c38349: RTP TS 1725021880 PC 1560319321 OC 3364521432 NTP TS 1121433750/3398500622=1121433750.791275 [1557513287.087290] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 9ba3be3b: FL 18 TL 4080170 HSR 3714308681 J 2085763389 LSR 1453166610 DLSR 3867570679 [1557513287.087303] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 9ba3be3b) - deleting SSRC 77b8933 [1557513287.087311] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513287.087324] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: Calling handler for RTCP packet type 201 [1557513287.087328] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: RR from 77c38349 about 477b6daa: FL 136 TL 7495912 HSR 3723043622 J 76729872 LSR 791112841 DLSR 2264873880 [1557513287.087332] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: SSRC hash table exceeded size limit (trying to add 477b6daa) - deleting SSRC b7800717 [1557513287.087338] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20053]: No payload type known for RTCP RR, discarding [1557513287.165301] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 200 [1557513287.165344] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SR from 779d978b: RTP TS 4203649664 PC 1900914033 OC 1842228915 NTP TS 2577169143/1958584749=2577169143.456018 [1557513287.165355] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 643dc91f: FL 113 TL 8548413 HSR 913668020 J 2977588542 LSR 2353716277 DLSR 1468396047 [1557513287.165370] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add 643dc91f) - deleting SSRC e03bd142 [1557513287.165382] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513287.165396] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: Calling handler for RTCP packet type 201 [1557513287.165403] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: RR from 779d978b about 7d2457ab: FL 45 TL 2024736 HSR 174032952 J 3137856066 LSR 2564654212 DLSR 844577710 [1557513287.165410] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: SSRC hash table exceeded size limit (trying to add 7d2457ab) - deleting SSRC 476ec510 [1557513287.165418] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340 port 20065]: No payload type known for RTCP RR, discarding [1557513316.000248] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Call branch 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00' (via-branch '') deleted, no more branches remaining [1557513316.000290] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Final packet stats: [1557513316.000297] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --- Tag '2684eff8-3602-41ed-aca2-9f7938362462', created 1:25 ago for branch '', in dialogue with 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00' [1557513316.000308] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: ------ Media #1 (audio over UDP/TLS/RTP/SAVP) using iLBC/8000 [1557513316.000318] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20052 <> 97.47.70.223:4667 , SSRC 77c38349, 55 p, 3410 b, 0 e, 32 ts [1557513316.000325] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20053 <> 97.47.70.223:4640 (RTCP), SSRC 77c38349, 12 p, 1120 b, 0 e, 29 ts [1557513316.000330] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --- Tag 'c71ac9de-9bf5-4f39-9fda-6bd3aaf97d00', created 1:25 ago for branch '', in dialogue with '2684eff8-3602-41ed-aca2-9f7938362462' [1557513316.000339] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: ------ Media #1 (audio over UDP/TLS/RTP/SAVP) using iLBC/8000 [1557513316.000347] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20064 <> 174.204.14.29:10523, SSRC 779d978b, 595 p, 40090 b, 0 e, 30 ts [1557513316.000355] INFO: [05c7dcfe-06f3-4630-9799-0557b76cc340]: --------- Port 209.182.216.71:20065 <> 174.204.14.29:10496 (RTCP), SSRC 779d978b, 17 p, 1490 b, 0 e, 29 ts [1557513316.000521] DEBUG: [05c7dcfe-06f3-4630-9799-0557b76cc340]: Sending DTLS packet

2 1

DMQ re-ordering concern
by Julien Chavanton 11 May '19

11 May '19

Yesterday at Kamailio World there was an interesting question, raising a concern about replication of user location and DMQ in general, we should track it in the mailing list and use it as a feature request and I can share my initial thoughts on the topic. The concern was about DMQ reordering transactions. With SIP registration this seems like a minor concern, high responsiveness at the cost of potential small inconsistency seems like a good option. Considering that, the state will automatically correct itself and that this can be controlled using the expiration timer. This will become even more insignificant when the replicas can only be used as primary server failover. (it becomes a very small concern only when the primary server fails) With other type of data re-ordering may result in more problematic use cases. My impression is that DMQ should try to be good only with volatile data, data that will expire anyway, like caching with expiration, registration, Dialogs, etc. I think it may be worth looking at adding support for ordering/queuing in DMQ. Performance and simplicity could be maintained by doing mainly transactions in parallel with a configurable re-ordering best effort to minimize the impact of the problem. For example : 1- trying to re-order for a configurable amount of time. (32 seconds to match the RFC for SIP re-transmissions) 2- adding tractability for failed transactions. 3- the possibility to trigger a re-sync. Another option is to look at preserving strict ordering but I can imagine this could add more problems in some cases ...

2 4

Question regarding NAPTR and RURI
by Schneuwly Mathias RUAG 10 May '19

10 May '19

Hi I've a question about NAPTR lockup and rewriting the RURI. Following a little example: - Alice at location A tries to call Bob at location B - location A uses the RURI naptr.locationB.com to reach location B - this NAPTR resolves to hostb.locationB.com and host B is listening on port 5070 hostb has the following alias defined: - udp:hostb.locationB.com:5070 The problem now is, that the incoming RURI is "sip:bob@naptr.locationB.com" and the check to myself is not True. Is there a possibility to rewrite the RURI to the result of the NAPTR query? If I would add a second alias like "udp:naptr.locationB.com:5070" a check to myself would still fail as the RURI does not contain a port. So I would have to add an alias like "udp:naptr.locationB.com" without port, then it would work.? Is this the way to go? Because the documentation says it is important to include the port, otherwise the loose_route() will not work as expected... Thanks for your replies! Mathias

1 0

Help identifying mem leaks
by Duarte Rocha 10 May '19

10 May '19

Greetings, Recently one of my Kamailio's started started answering with a lot of 500 replies to requests because of insufficient private memory. The service Kamailio had been running for 6 months now. In order to solve this i increased the private memory settings and restarted the Kamailio. Before that, when searching a bit about the issue i found an old thread where Daniel suggested using " kamcmd corex.pkg_summary idx 1" to check for pkg leaks in the sip worker process. I ran this before doing the restart in order to identify leaks and found the summary in the logs. However, i am having a hard time interpreting this summary and being able to know if i have or not a mem leak. What should i be looking for? This is the beggining of the summary : qm_status: (0x7ff3c8cef010): qm_status: heap size= 8388608 qm_status: used= 928576, used+overhead=1464080, free=6924528 qm_status: max used (+overhead)= 1488472 qm_status: dumping all alloc'ed. fragments: qm_status: 0. N address=0x7ff3c8d286d8 frag=0x7ff3c8d286a0 size=1024 used=1 qm_status: alloc'd from core: core/str_hash.h: str_hash_alloc(59) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 1. N address=0x7ff3c8d28b40 frag=0x7ff3c8d28b08 size=256 used=1 qm_status: alloc'd from core: core/str_hash.h: str_hash_alloc(59) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed qm_status: 2. N address=0x7ff3c8d28ca8 frag=0x7ff3c8d28c70 size=184 used=1 qm_status: alloc'd from core: core/counters.c: cnt_hash_add(332) qm_status: start check=f0f0f0f0, end check= c0c0c0c0, abcdefed Any help? Best Regards, Duarte Rocha

1 0

How to configure Kamailio 5.1 with Asterisk pjsip
by przeqpiciel 10 May '19

10 May '19

Hi, There is any working examples how to configure dispatcher module of Kamailio? I have working registrations but working in half dialing function.Signalling are come to asterisk and rtp start sending to my IP but i cant hear anything. Asterisk are doing registrar job - i dont know if it is good idea, maybe not.

5 11

RTP multiplexing, batching and compression (KamailioWorld followup)
by Stefan Sayer 09 May '19

09 May '19

Hi, as a followup to the 5 min 5 slides talk on Tuesday at the Kamailio world conference, I wanted to send links to that implementation, as I did get some interest about this afterwards. This transparent RTP mux gateway implementation in SEMS is a multiplexing, batching and compression mechanism, which can save quite some bandwidth on low bitrate codecs - can be useful if you need to work with very low bitrate links, e.g. on satellite (I tested with 10 G729 calls instead of 5 on a 128kbit sat link). It's a bit like ROHC (RFC3095) but on application level, only for RTP, and by using configuration for IP addresses instead of negotiation. Or as osmux, but without separate negotiation (but in-band), and for all codecs, and SIP. Or iax2 trunking mode, but with normal SIP/RTP. https://github.com/sems-server/sems/blob/sayer/rtp-mux/doc/rtp_mux/rtp_mux_… https://github.com/sems-server/sems/blob/sayer/rtp-mux/doc/rtp_mux/rtp_mux.… For now it's in the sayer/rtp-mux branch, but it's planned to merge it to master, possibly with simplification of the configuration. On another note, many thanks to the organizers of the Kamailio World conference! It's both really nice to meet, and very useful, and I think as well individually as well as for the community as a whole. Best Regards, Stefan

1 0

CRM / Windows integration
by Mark Boyce 09 May '19

09 May '19

Hi All Does anyone have any pointers for integrating Kamailio with CRMs (Dynamics / Salesforce / Zoho) or native Windows? We’re looking for basic click to dial / screen pop. Thanks Mark -- Mark Boyce Dark Origins Ltd

4 5

RE-invite in case of SIP is Active
by Gaurav Bmotra 09 May '19

09 May '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 How can i use Re-invite in kamailio if SIP is active so that call can reconnect in case of network loss plz help -- *Regards:* Gaurav Kumar

2 2

Call reconnect in case of loss network
by Gaurav Bmotra 09 May '19

09 May '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 when i call from one to another extension , in case of loss network it disconnect the call or stuck there , how can i make it as re-connection when network loss , and it will try to reconnect for some seconds like 30 sec "Example for callingn skype it reconnectcall in case of lossnetwork " plz help -- *Regards:* Gaurav Kumar

1 0

Sending in-dialog requests from custom module's C code
by Ivan Ribakov 09 May '19

09 May '19

I’m working on a custom Kamailio module where I need to send new-dialog and in-dialog requests on timer (hence I’m being forced to generate new messages from C code). So far I’ve been using modules/tm/tm_load.h defined interface to generate messages and handle callbacks. New dialog messages are sent and processed normally. To do that I’m: 1. Calling set_uac_request() to define request parameters 2. Calling tmb.t_request_outside() to send it outside any existing dialog I was able to send in-dialog requests (or so I thought) in a similar fashion, but I soon realised that responses to those requests were dropped because they couldn’t be matched against any known existing transaction. I’m attaching log messages that I believe support this theory and I’ve also observed UDP retransmissions. tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 50576 label 0 branch 0 tm [t_lookup.c:990]: t_reply_matching(): no matching transaction exists tm [t_lookup.c:993]: t_reply_matching(): failure to match a transaction tm [t_lookup.c:1088]: t_check_msg(): msg (0x5abcb50) id=1 global id=1 T end=(nil) tm [t_reply.c:2195]: reply_received(): transaction not found - (branch -1) The way I’m currently generating in-dialog requests is very similar to what tmb.t_request_outside() does, the main difference being that I do the dialog setup manually, based on the call-ID, cseq and from/to tags (I’m tracking transaction identifiers separately) and then pass resulting uac_req_t to tmb.t_request_within() - https://gist.github.com/IvanRibakov/3302cb286b1f4b786d109b406f2435a2 <https://gist.github.com/IvanRibakov/3302cb286b1f4b786d109b406f2435a2> Now, the question part - does anyone know what I’m doing wrong/missing? As I mentioned, when looking at the generated message bodies, they look ok to me (left - initial request that started the dialog, right - first in-dialog request), so I’m guessing I’m missing some Kamailio internal steps needed to register new transaction. SIP flow (up to the point when first UDP retransmission happens) I apologise in advance for the bulky question and will be extremely thankful for any guidance. Regards, Ivan

2 2

GDB debugging Kamailio 5.2.2 IMS authorization failed by the SCSCF
by Woscek, Martin W. 08 May '19

08 May '19

Hi, Looking for some advice running the debugger and setting breakpoints in Kamailio modules such as the SCSCF. Example: We were trying to watch the authenticate() function: 714 int authenticate(struct sip_msg* msg, char* _realm, char* str2, int is_proxy_auth) { Our Kamailio 5.2.2 code was compiled and linked from source with the -g option already enabled by default. We also dropped our Workers to 1 from 16 in the scscf.xml file for the SCSCF to limit the thread count and get the following list of threads below: We tried the first cdp threads: 8462, 8463, and 8464 and while we could set a break point at line 714 for the authenticate() function, it didn't actually break there at all. Question: What pid will we use to catch the authenticate() called during an initial registration from a boghe client? Anything special we may be missing, gdb related or Kamailio settings? kamcmd -s /run/kamailio_scscf/kamailio_ctl kamcmd 1.5 Copyright 2006 iptelorg GmbH This is free software with ABSOLUTELY NO WARRANTY. For details type `warranty'. kamcmd> ps 8456 main process - attendant 8457 udp receiver child=0 sock=10.0.20.1:6060 8458 slow timer 8459 timer 8460 secondary timer 8461 ctl handler 8462 cdp worker child=0 8463 cdp receiver peer unknown 8464 cdp_receiver_peer=hss522.irisims.org 8465 cdp_acceptor 8466 cdp_timer 8470 notification_worker 8471 notification_worker 8472 notification_worker 8473 notification_worker 8477 JSONRPCS FIFO 8478 JSONRPCS DATAGRAM Any advice appreciated. Thanks, Martin Woscek

2 1

http_client_query module?
by Andrew Chen 08 May '19

08 May '19

Hey guys, Need help here. What module besides tls.so I need to load in order to get http_client_query to work? The module doc says tls.so: #!ifdef WITH_TLS loadmodule "tls.so" #!endif but I keep getting this when doing config check: root@ashmaintpsg51:/etc/kamailio# kamailio -c kamailio.cfg 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 322, column 71: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 323, column 45: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 324, column 42: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 325, column 48: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 326, column 44: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 327, column 71: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 328, column 67: Can't set module parameter 0(12971) ERROR: <core> [core/modparam.c:152]: set_mod_param_regex(): No module matching <http_client> found 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 329, column 41: Can't set module parameter 0(12971) ERROR: <core> [core/cfg.y:3309]: yyparse(): cfg. parser: failed to find command http_client_query (params 2) 0(12971) CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 859, column 105: unknown command, missing loadmodule? and I have the package installed on my ubuntu 18.04: ii kamailio-*tls*-modules:amd64 5.1.2-1ubuntu2 amd64 TLS support for the Kamailio SIP server (authentication, transport) thoughts? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 3

Udp errors on High Availability Setup
by Duarte Rocha 08 May '19

08 May '19

Greetings, I have two Machines with Kamailio in a High Availability (HA) configuration. Both machines are active and processing calls and both have one HA IP that can jump to the other machine in case of failure. I also have probing for NAPs configured with each IP as the socket. So, in each machine i have this configuration : listen=udp:HA_Address1 listen=udp:HA_Address2 NAP 1 being probed from socket HA_Address1 NAP 1 being probed from socket HA_Address2 I also have /proc/sys/net/ipv4/ip_nonlocal_bind with value 1. Everything works correctly with this configuration. However when the probing is sent for the NAP' my logs get filled with the same error messages. udp_send(): sendto(sock, buf: 0x7f2d4c91fa18, len: 340, 0, dst: (xxx.xxx.xxx.xxx:xxx), tolen: 16) - err: Invalid argument (22) CRITICAL: <core> [core/udp_server.c:602]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed The error happens because it is trying to probe a NAP using a socket that isn't active at the moment in the machine. Is there a way for this error not to be logged or be disabled without reducing Log Level to suppress CRITICAL errors? I think this is a standard High Availability configuration. Is there some setting that can disable this kind of errors? Best Regards, Duarte Rocha

3 5

HELP ME
by khadime gaye 07 May '19

07 May '19

Good morning, How I can resolve this problem ? << ERROR: Could not load the script in /usr/local/lib64/kamailio//kamctl/kamdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' >>

2 1

implementing custom authentication method
by Walter.Martín Villalba 03 May '19

03 May '19

Dear list, I'm implementing a custom method for user authentication and would like to kindly ask for some guidance regarding the server side changes that may be needed in order to keep things running smoothly. The idea is to replace the standard user/pass-based digest authentication with a custom one based on Open ID Connect (OIDC) access tokens. I already have some pieces working (receiving access token in REGISTER message, validating the token by calling custom python routine, etc.), but I'm missing the bits needed to keep track of users who have successfully authenticated. While looking at sipcomm.cfg I found calls to *www_authenticate* (defined in modules/auth_db/authorize.c), which in turn calls to *mark_authorized_cred*, *check_auth_hr* (or *auth_check_hdr_md5*), and *generate_avps*,. Some of these functions are indeed changing some values in the DB and/or in memory, and this is where I would like some guidance. Is there any documentation I could look at which hopefully explains what DB values and/or data structures in memory need to be updated in order to keep track of successfully authenticated users? I believe with this last piece I'd be able to wrap up my custom authentication implementation. Thanks in advance! Regards, Walter Martín.

1 0

Kamailio bypassing RTP media
by Ali Taher 03 May '19

03 May '19

Hi everyone, I'm new to kamailio. I plan to setup a sip server cluster, Does someone can give me some suggestions if I can use kamailio as front server, which handle sip message and bypass rtp media message to backendserver, this mean the kamailio handle only sip message, the rtp packets are sent directed to backend server. The sip sequences is 1.softswitch send invite to kamailio 2.kamailio send invite to backend server. 3.backend server return 200 ok to kamailio 4.kamailio return 200 ok to softswitch, the ip and port in sip is from the backend server's msg, this make the softswitch send rtp direct to backend server. 5.rtp packets are transfered between softswitch and backend server. .... 6.backend server send bye to kamailio 7.kamailio send bye to softswitch. Any help is appreciated. Regards, Ali Taher

4 5

forcing socket doesn't work for ACK
by David Villasmil 03 May '19

03 May '19

Hello guys, I have public and private IPs, and i need to force the sending socket replying to internal out externals endpoint, so i'm using this: function to check the destination (i tried using $sndto(ip) but it always returns NULL): route[CHECK_DEST_NET] { if ( $du=~"sip:172\..*" ) { setflag(FLAG_TO_PRIVATE); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PRIVATE -> [$du]\n" ); } else { setflag(FLAG_TO_PUBLIC); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PUBLIC -> [$du]\n" ); } return; } on my relay: route(CHECK_DEST_NET); if (isflagset(FLAG_TO_PRIVATE)) { xlog("L_ERR", "[RELAY] forcing socket to PRIVATE NET\n" ); force_send_socket(LISTEN_INSIDE_IF:LISTEN_INSIDE_PORT); } else { xlog("L_ERR", "[RELAY] forcing socket to PUBLIC NET\n" ); force_send_socket(LISTEN_OUTSIDE_IF:LISTEN_OUTSIDE_PORT); } This works nicely, except for ACKs and potentially other packets sent statelessly? Anyone knows how to fix this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

6 26

run some extra things before relying message
by Mojtaba 02 May '19

02 May '19

Hello, I have encountered with problem in Kamailio. At first glance it looks simple. Considering the Data Lump feature in Kamailio, All changes in SIP body, does not apply quickly. So if we are using RTPEngine or RTPProxy, How we can get body message (Actually IP and port from SDP after rtpengine_manage function) before relaying message? Suppose i want get some information and run some extra things before relying message? I just added my extra things in t_on_reply block, but it's strange why it doesn't work before relaying! -Mojtaba Esfandiari.S

2 2

Problem with topoh
by Phillman25 Kyriacou 02 May '19

02 May '19

Dear List I have an issue with topoh module. Specifically for in-dialog requests, if the encoded uri is altered i.e. initial encoded contact uri is Contact: <sip:X.X.X.X;line=sr-huAWi7oPBYsF.cC0L2lWOUlPwbSWOUl6wURTw24W> and UAS sends an in-dialog INVITE like this adding tgrp and user=phone: INVITE sip:X.X.X.X;line=sr-huAWi7oPBYsF.cC0L2lWOUlPwbSWOUl6wURTw24W;tgrp=1069912311;user=phone Kamailio cant seem to match/decode this and responds "404 Not Found" I have also set: modparam("topoh", "uri_prefix_checks", 0) As specified here: https://www.kamailio.org/docs/modules/4.4.x/modules/topoh.html#idp37582868 But still doesn't work. Is there a bug with this module? # ==================================================== # ================ TOPOLOGY HIDING =================== # ==================================================== modparam("topoh", "mask_key", "secret") modparam("topoh", "mask_ip", "X.X.X.X") modparam("topoh", "mask_callid", 0) modparam("topoh", "uparam_name", "line") modparam("topoh", "uparam_prefix", "sr-") modparam("topoh", "vparam_name", "branch") modparam("topoh", "vparam_prefix", "z9hG4bKsr-") modparam("topoh", "uri_prefix_checks", 0) Thanks in advance for your support Phillip

2 4

Stack smash error when using auth_ephemeral with KEMI
by Grant Bagdasarian 02 May '19

02 May '19

Hello, I was trying out the auth_ephemeral module inside the xhttp event_route in Python KEMI, and discovered a possible bug. https://pastebin.com/4Ghumfis The following will result in the stack smash error. Whenever I leave out the unixtimestamp the module will complain about not being able to convert to an int, which make sense, but doesn't lead to a stack smash error and a complete crash of Kamailio. KSR.auth_ephemeral.autheph_authenticate("1576734089:grant", "1234"). According to the documentation this module can only be used in REQUEST_ROUTE, but I found a presentation online which also used this module and that particular function in event_route, which is what I need for WebSocket authentication. Hope this helps. Please let me know if you need more information. Regards, Grant

2 1

Kamailio 5.2.2 IMS authorization failed by the SCSCF
by Woscek, Martin W. 02 May '19

02 May '19

Hi, We are getting a SIP 401 Unauthorized -Challenging the UE error originating by the SCSCF. It looks to be related to the nonce/cnonce. Is there something we have misconfigured in our SCSCF module? Log output: May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18489]: DEBUG: ims_auth [authorize.c:786]: authenticate(): Nonce or response missing: nonce len [-108158976], response16 len[1873606560] May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [utils.c:168]: get_nonce_response(): Calling find_credentials with realm [ims522.irisims.org] May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [utils.c:195]: get_nonce_response(): Found nonce response May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: INFO: ims_auth [authorize.c:827]: authenticate(): uri=sip:irisims.org nonce=9863c63855c8dd888c5c18e79fc61156 response=ce7ce73872b92aff2defa76dd1aff1ac qop=auth-int nc=00000001 cnonce=4fe4daacd69fa3d609c8b396d331eae6 hbody=d41d8cd98f00b204e9800998ecf8427e May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [rfc2617.c:135]: calc_response(): calc_response(_ha1=7a1b479a0513914e71bcd6d005b3e17d, _nonce=9863c63855c8dd888c5c18e79fc61156, _nc=00000001,_cnonce=4fe4daacd69fa3d609c8b396d331eae6, _qop=auth-int, _auth_int=1,_method=REGISTER,_uri=sip:irisims.org,_hentity=d41d8cd98f00b204e9800998ecf8427e) Thanks, Martin

2 1

New voisonics.com - Kamailio related products and services
by David Cunningham 02 May '19

02 May '19

Hello, Usually it's our client's VoIP technology being updated. Fortunately we found a little spare time to completely revamp the Voisonics website as well! https://voisonics.com/ Voisonics provides products and services in the commercial VoIP and hosted PBX sector; in particular relating to Kamailio, Asterisk, and Enswitch. The business has been specialising in this area for almost 10 years. Please feel free to contact me off-list with any questions. Kind regards, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

1 0

Support for JWT token authentication?
by David Dean 01 May '19

01 May '19

I came across this thread from Jan 2015 discussing the addition of JWT token authentication to Kamailio: https://github.com/kamailio/kamailio/issues/29 Was JWT authentication ever added? Or has anyone else on this list got JWT working somehow?

1 0

CSeq not incrementing - app_ruby
by Andrew White 01 May '19

01 May '19

Hi all, I’m still building with app_ruby and loving it - for the most part! I’m having an issue with one of my providers responding with a 482 on auth invite. Researching, it appears that my CSeq isn’t incrementing and this is the reason for the issue. I already have the relevant flag set in my kamailio.cfg however: modparam("dialog", "track_cseq_updates", 1) During my uac_auth, I’ve tried manually setting the diff, as well as running msg_iflag_reset (as suggested by Daniel - https://github.com/kamailio/kamailio/issues/1359 <https://github.com/kamailio/kamailio/issues/1359> - unfortunately the function isn’t exported in KEMI): if KSR::UAC.uac_auth() then KSR.info("UAC authed, relaying") #KSR::COREX.msg_iflag_reset("UAC_AUTH") KSR::PV.sets("$dlg_var(cseq_diff)", "1") KSR.info("CSeq diff: #{KSR::PV.gete("$dlg_var(cseq_diff)")}") KSR::TM.t_relay() Unfortunately in both cases the actual CSeq doesn’t increment in the second INVITE: INVITE 1 (unauthenticated): > CSeq: 102 INVITE INVITE 2 (with Authorization header): > CSeq: 102 INVITE I’m unsure if I’ve missed something silly around the track_cseq_updates flag, or if app_ruby somehow isn’t interacting with the dialog module when running uac_auth to trigger the increment? Thanks! Kind regards, Andrew

3 6

listening at link local IPv6 address
by Juha Heinanen 01 May '19

01 May '19

My K host is behind a mobile WiFi hotspot and I tried to make K listen at the IPv6 address fe80::6e29:95ff:fe7d:37e6/64 that the host has received from the hotspot. Using that address on listen line gives error: Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:395]: udp_init(): bind(8, 0x7ff97dfb3fe4, 28) on FE80:0:0:0:6E29:95FF:FE7D:37E6: Invalid argument Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:401]: udp_init(): might be caused by using a link local address, try site local or global I can ping that address like this: $ ping6 fe80::6e29:95ff:fe7d:37e6%wlp1s0 PING fe80::6e29:95ff:fe7d:37e6%wlp1s0(fe80::6e29:95ff:fe7d:37e6%wlp1s0) 56 data bytes 64 bytes from fe80::6e29:95ff:fe7d:37e6%wlp1s0: icmp_seq=1 ttl=64 time=0.030 ms I then tried to add %wlp1s0 suffix to the listen address, but got error: Apr 21 09:10:35 salmon sip-proxy[7896]: 0(7948) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 408, column 34: ip address, interface name or hostname expected Did I do something wrong? If not, why is it that K cannot listen at a link local address? It is something that Linux networking stack does not support? -- Juha

6 33

Pipelimit with clean_unused
by Cinthia Leung 30 Apr '19

30 Apr '19

Hello, What is your recommendation on using clean_unused with db defined pipes? >>pl_check("systemwide"); If I leave the system idle, they are eventually "cleaned" as expected. In our use case, that means failed calls unfortunately. I am trying to see if I can mix both db/static pipes with dynamic/run-time pipes. Thanks Cindy

1 1

Next hop of reply, in onreply_route
by Alex Balashov 30 Apr '19

30 Apr '19

I am fairly sure it’s been asked a number of times, but is there a way to know the destination, and more especially the transport, of the next hop of a reply message, viewable from onreply_route? For requests, we have $nh and many other PVs which in some way evidence where it’s going. And yes, I’m aware that this can be previewed from the onsend_routes*, but I need to amalgamate that knowledge with some transaction-specific data. Thanks! — Sent from mobile, with due apologies for brevity and errors.

2 2

"ERROR: reply too big" and ctl binrpc size
by Karsten Horsmann 30 Apr '19

30 Apr '19

Hi all, i use kamailio 5.0.7 in production and i noticed that i cant listen the dialogs after a few days of traffic on this server. Then i got "kamcmd dlg.list_ctx " - "ERROR: reply too big". So now my question - i want to debug my "sleeping dialogs" with dlg.list - how big should it change the values of without knowing the size of the current filled up binrpc stuff? Or is there a better option to debug dialog-module? modparam("ctl", "binrpc_max_body_size", ??) modparam("ctl", "binrpc_struct_max_body_size", ?) -- Cheers and see you in Berlin *Karsten Horsmann*

2 1

Re: [SR-Users] SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi Daniel, Thank you very much for clarifying! Cheers, Yufei On 30 Apr 2019 at 08:10, <Daniel-Constantin Mierla <miconda(a)gmail.com>> wrote: Hello, there is an errata for that RFC, but it refers only to the BYE, where it adds back the transport=tcp, but the ACK R-URI should follow the same rules, see: * https://www.rfc-editor.org/errata/eid5294 Cheers, Daniel On 28.04.19 22:54, Yufei Tao wrote: Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei _______________________________________________ Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

docker image for non x86 arch
by Sergey Safarov 30 Apr '19

30 Apr '19

Is the community involved in kamailio docker image for ARM architecture and others not x86 platforms? ARM ARM 64 IBM POWER IBM Z PowerPC 64 LE

3 3

SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei

3 2

remove_record_route - bug?
by Sergiu Pojoga 29 Apr '19

29 Apr '19

Hi ppl, After a branch route fails, I need to re-evaluate the next destination and adjust RR parameters, if need be, like for example adjust the nat=yes|priv. failure_route[MANAGE_PSTN_FAILURE] { ... remove_record_route(); record_route(); route(NATMANAGE); ... route(RELAY); } The result is that the new branch is stripped of "Record-Route: <sip:MYIP>", but the params remained. Moreover, record_route hasn't been re-added as instructed in the failure route. Request headers of the initial branch that will eventually timeout and fail: 2019/04/27 18:29:18.034992 10.22.0.1:5060 -> 10.22.0.100:5060 INVITE sip:1514XXXXXXX@10.22.0.100 SIP/2.0 Record-Route: <sip:10.22.0.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> Record-Route: <sip:65.XX.XX.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> New request branch after failure: 2019/04/27 18:29:19.538292 65.XX.XXX.1:5060 -> 65.XX.XX.2:5060 INVITE sip:1514XXXXXXX@65.XX.XX.2 SIP/2.0 ;lr=on;did=9e2.3dd2;nat=priv> kamailio -v version: kamailio 5.1.6 (x86_64/linux) 7d1964 Much obliged. --Sergiu

4 5

Vide SIP conferencing kamailio with freeswitch
by Gaurav Bmotra 28 Apr '19

28 Apr '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help -- *Regards:* Gaurav Kumar

2 1

Re: [SR-Users] [sr-dev] Vide SIP conferencing kamailio with freeswitch
by Henning Westerholt 28 Apr '19

28 Apr '19

Hello, please ask this usage related questions on our "sr-users" list. Kamailio is not doing much regarding the video calls. E.g. if your calls work for voice with kamailio, they should also work for video. All the setup needs to be done in freeswitch. I found a (outdated) howto with google: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video Maybe this helps. Henning Am 27.04.19 um 13:22 schrieb Gaurav Bmotra: HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

1 0

How to configure ipv6 in kamailio
by Santosh Kumar Dash 28 Apr '19

28 Apr '19

Hi All, I have a kamailio installed in my setup which is working properly in IPV4 configuration. I want to perform IPV6 configuration, I don't have any idea how to configure it, I have tried few things by changing *kamailio.cfg* file but did not work. I am sharing my *kamailio.cfg *file which is working fine in ipv4 Kindly help me to configure for IPV6. lets say my Ipv6 address is : 2402:3a80:d1a:f45a:fe44:2e54:ed2b:9bb2 Thanks,

3 3

Can't set module parameter
by Tmbc 26 Apr '19

26 Apr '19

Hi all Having this issue with kamailio: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 119, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <grp_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 120, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <cnt_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 121, column 50: Can't set module parameter here is full conf file: https://dpaste.de/bP3E OS: NAME="CentOS Linux" VERSION="7 (Core)" Software version: kamailio 5.2.2 (x86_64/linux) 67f967 Thanks in advance, tmbc

2 1

Re: [SR-Users] Config to control TCP connection reuse
by R, Rajkumar (Raj) 24 Apr '19

24 Apr '19

I think elaborating on the setup may help someone to provide inputs on this. Below is the setup. Client 1 --> Kamailio proxy --> Backend SIP server Client 2 --> Kamailio proxy --> Backend SIP server Client 3 --> Kamailio proxy --> Backend SIP server After I register all three clients, see only one TCP connection is being used between "Kamailio proxy - Backend SIP server". Is there a way to achieve separate TCP connection to Backend SIP server from kamailio. thanks, raj From: R, Rajkumar (Raj) Sent: Monday, April 22, 2019 3:45 PM To: sr-users(a)lists.kamailio.org Subject: Config to control TCP connection reuse Hi, In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one. Could anyone help how to achieve different TCP connections to same destination in kamailio? thanks, raj

2 1

DMQ_USRLOC
by Gholamreza Sabery 24 Apr '19

24 Apr '19

Hi, Recently, I started to use dmq_usrloc module. I successfully set it up. Unfortunately, the documentation does not specify the behavior of dmq_usrloc module with other Kamailio modules. For example, I see a bunch of KDMQ messages are exchanged between nodes, but how the lookup() function behaves with dmq_usrloc, or what is it's effect on a db_mode of 3! Regards

5 6

JSON error - control character 0xd
by David Dean 23 Apr '19

23 Apr '19

I'm trying to send an SDP from a SIP client to Janus using JSON over HTTP. The problem is that Janus is erroring on the SDP content with the following error: JANUS HTTP: Get SDP for echo plugin: {#012 "janus": "error",#012 "error": {#012 "code": 454,#012 "reason": "JSON error: on line 9: control character 0xd near '\"v=0'"#012 }#012} Result code 200 I tried using {s.escape.common} to escape any quotations, commas etc but it doesn't appear to replace the control characters with newlines. Any suggestions on how best to remove these control characters so that Janus will accept the SDP over JSON? Here is the relevant section of kamailio.cfg: # HTTP: Request SDP for echo test plugin sdp_get("$avp(sdp)"); $var(res) = http_connect("janus", "/janus/$var(JANUS-ID)/$var(ECHO-ID)", "application/json", "{ \"janus\" : \"message\", \"transaction\" : \"testtesttest99\", \"body\" : { \"audio\" : true }, \"jsep\" : { \"type\" : \"offer\", \"sdp\" : \"$(avp(sdp){s.escape.common})\" }}", "$avp(janus-pluginsession)"); xlog("L_INFO", "JANUS HTTP: Get SDP for echo plugin: $avp(janus-pluginsession) Result code $var(res)\n");

3 6

Direction of replies
by Duarte Rocha 23 Apr '19

23 Apr '19

Greetings, Is there a way to know in reply_route if a reply has direction downstream or upstream regarding the first INVITE? I know there are vars and methods for finding this out on requests, but i can't find anything for replies. Thanks in advance, Best Regards, Duarte Rocha

2 1

CDR start time of a canceled call
by Jan-Hendrik Dörner 23 Apr '19

23 Apr '19

Hello, I just noticed that the logged „start_time" as well as „end_time" of a canceled call is set to the „invite time“. If done this way, it is harder to calculate the „ringing time“. Is there an option to change this behaviour (i.e. to log the time of the received „bye“ instead)? (If done so, the ringing time would always be the time between start and invite …) Or is there a special reason for this behaviour? My CDR settings are as followed: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) # log unanswered / open dialogs modparam("acc", "cdr_on_failed", 1) # log also not answered calls modparam("acc", "cdr_extra", "inviteTime=$dlg_var(inviteTime)" modparam("acc", "cdr_log_enable", 1) # activate write to syslog modparam("acc", "cdr_facility", "LOG_LOCAL6") # syslog modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "time_mode", 0) #syslog timestamp modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) # needed for syslog modparam("acc", "log_missed_flag", FLT_ACCMISSED) # need for logging missed calls in syslog Regards Jan-Hendrik

1 0

Kamailio-5.2.1 - Issues in NDB_MONGODB module initialisation
by José Seabra 22 Apr '19

22 Apr '19

Hello there, I have NDB_MONGODB loaded in kamailio with following configurations: *loadmodule "ndb_mongodb"* *modparam("ndb_mongodb", "server", "name=mgs1;uri='mongodb://mongodbserver:27017/proxyregistrar'")* Kamailio breaks up on start up and prints the following messages: *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/modparam.c:83]: set_mod_param_regex(): 'ndb_mongodb' matches module 'ndb_mongodb'* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/sr_module.c:711]: find_param_export(): found <server> in module ndb_mongodb [/usr/lib64/kamailio/modules/ndb_mongodb.so]* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/modparam.c:99]: set_mod_param_regex(): found <server> in module ndb_mongodb [/usr/lib64/kamailio/modules/ndb_mongodb.so]* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: ndb_mongodb [mongodb_client.c:155]: mongodbc_add_server(): added server[mgs1]=mongodb://10.225.121.12:27017/proxyregistrar <http://10.225.121.12:27017/proxyregistrar>* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/sr_module.c:942]: init_mod_child(): idx 21 rank 8: ndb_mongodb [tcp receiver (generic) child=3]* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 23* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: ALERT: <core> [main.c:739]: handle_sigs(): child process 3181 exited by a signal 11* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: ALERT: <core> [main.c:742]: handle_sigs(): core was generated* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/tcp_main.c:3513]: handle_ser_child(): dead child 16, pid 3181 (shutting down?)* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: INFO: <core> [main.c:764]: handle_sigs(): terminating due to SIGCHLD* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa6d0e0, 23, -1, 0x0) fd_no=23 called* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [main.c:766]: handle_sigs(): terminating due to SIGCHLD* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: ndb_mongodb [ndb_mongodb_mod.c:128]: mod_destroy(): cleaning up * Core dump back trace output: *(gdb) bt full* *#0 0x00007f3daa134f93 in _mongoc_client_new_from_uri () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#1 0x00007f3daa156553 in mongoc_uri_get_ssl () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#2 0x00007f3daa134f88 in _mongoc_client_new_from_uri () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#3 0x00007f3daa135029 in mongoc_client_new () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#4 0x00007f3daae6d75f in mongodbc_init () at mongodb_client.c:63* * rsrv = 0x7f3db0f86728* * __FUNCTION__ = "mongodbc_init"* *#5 0x00007f3daae73c46 in child_init (rank=3) at ndb_mongodb_mod.c:113* * __FUNCTION__ = "child_init"* *#6 0x00000000005470d7 in init_mod_child (m=0x7f3db0f85e88, rank=3) at core/sr_module.c:846* * __FUNCTION__ = "init_mod_child"* *#7 0x00000000005474b6 in init_child (rank=3) at core/sr_module.c:874* * ret = 32573* *#8 0x00000000004b3840 in fork_process (child_id=3, desc=0x7ffecc20c7f0 "udp receiver child=2 sock=10.225.121.93:5060 <http://10.225.121.93:5060>", make_sock=1) at core/pt.c:341* * pid = 0* * child_process_no = 3* * ret = -1* * new_seed1 = 1273263324* * new_seed2 = 1531871627* * sockfd = {14, 15}* * __FUNCTION__ = "fork_process"* *#9 0x0000000000424b9b in main_loop () at main.c:1631* * i = 2* * pid = 3807* * si = 0x7f3db123a988* * si_desc = "udp receiver child=2 sock=10.225.121.93:5060 <http://10.225.121.93:5060>\000\177\000\000P\310 \314\376\177\000\000\000e|\261=\177\000\000 \313 \314\376\177\000\000\063\225{\000\000\000\000\000\027\000\000\000\000\000\000\000\000\000\000 \000\000\000\000\000\000\200\000\000\000\000\000\217e|\261=\177\000\000`~x\000\000\000\000\000\230\304\022\261=\177\000"* * nrprocs = 4* * woneinit = 1* * __FUNCTION__ = "main_loop"* *#10 0x000000000042c4e9 in main (argc=11, argv=0x7ffecc20cc08) at main.c:2675* * cfg_stream = 0x2893010* * c = -1* * r = 0* * tmp = 0x7ffecc20e6fd ""* * tmp_len = 0* * port = 0* * proto = 2496* * options = 0x7697a0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:"* * ret = -1* * seed = 173321635* * rfd = 4* * debug_save = 0* * debug_flag = 0* * dont_fork_cnt = 0* * n_lst = 0x7ffecc20cac0* * p = 0x7f3db177497d <intel_check_word+173> "H\205\300t\025\017\266P\003\017\266\312D9\371\017\204", <incomplete sequence \316>* * st = {st_dev = 19, st_ino = 21919, st_nlink = 2, st_mode = 16832, st_uid = 993, st_gid = 2, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1551179759, tv_nsec = 561093463}, st_mtim = {tv_sec = 1551179759,* *---Type <return> to continue, or q <return> to quit---* * tv_nsec = 561093463}, st_ctim = {tv_sec = 1551179759, tv_nsec = 561093463}, __unused = {0, 0, 0}}* * __FUNCTION__ = "main"* *(gdb)* NDB_MONGODB configuration is pretty simples and currently i'm not making any mongodb operations, i'm just loading the module and trying to connect against mongodb. Any thoughts on this issue? Thank you Best regards -- Cumprimentos José Seabra

2 1

rtpengine_offer() with ICE not updating o= and m= IP addresses
by David Dean 22 Apr '19

22 Apr '19

I'm using the following rtpengine_offer() to force the use of ICE relay and also replace o= and m= rtpengine_offer("replace-origin replace-session-connection ICE=force-relay RTP"); The SDP is being updated to include an ICE relay candidate, but the IP addresses in the o= and m= lines are not changing to the servers IP address (X.X.X.X). I also tried using the media-address parameter but that didn't change the behaviour. When I use rtpengine_answer() then it's not providing an ICE candidate nor changing o= or m=, which I imagine is expected. What am I missing? INVITE from the client (before rtpengine)------------------------------------------------------ INVITE sip:server.domain.com SIP/2.0Via: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rportFrom: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>Contact: <sip:username@154.20.1.8:37520;transport=tcp>Call-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Content-Length: 1506Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneSupported: 100rel --7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=ptime:20a=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=rtpmap:114 AMR-WB/16000/1a=imageattr:114 octet-align=0a=fmtp:114 octet-align=0a=rtpmap:115 AMR-WB/16000/1a=imageattr:115 octet-align=1a=fmtp:115 octet-align=1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:101 0-16a=pcfg:1 t=1a=sendrecva=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audio INVITE from the server (after rtpengine)------------------------------------------------------ SIP/2.0 200 OKVia: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rport=37520;received=2.221.73.230From: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>;tag=0ea7681ffd64b0ffcde1bff3393fd505.ae3aCall-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContact: <sip:X.X.X.X:443;transport=tcp>Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 1965 --7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=imageattr:114 octet-align=0a=imageattr:115 octet-align=1a=pcfg:1 t=1a=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audioa=rtpmap:114 AMR-WB/16000/1a=rtpmap:115 AMR-WB/16000/1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:114 octet-align=0a=fmtp:115 octet-align=1a=fmtp:101 0-16a=sendrecva=ptime:20a=ice-ufrag:kzgbFAnua=ice-pwd:hnEu9EwTrdIZnTWhYbxtmdugvra=candidate:dzkwmPcB28NPHfRA 1 UDP 16777215 X.X.X.X 30366 typ relay raddr X.X.X.X rport 30366a=candidate:dzkwmPcB28NPHfRA 2 UDP 16777214 X.X.X.X 30367 typ relay raddr X.X.X.X rport 30367a=sendrecva=ice-ufrag:uDE9wbvYa=ice-pwd:dx4CcIjqUNFfnRPJEF8GjatXdU

3 3

Config to control TCP connection reuse
by R, Rajkumar (Raj) 22 Apr '19

22 Apr '19

Hi, In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one. Could anyone help how to achieve different TCP connections to same destination in kamailio? thanks, raj

1 0

Best tutorial for beginners kamailio
by Gaurav Bmotra 22 Apr '19

22 Apr '19

Hi Plz sujjest best kamailio tutorial or any source where a beginner can start kamailio.. Thanks

2 1

push to talk over cellular application server and related media server implementation
by Thierry Luo 21 Apr '19

21 Apr '19

Hi All, I don't know if this is the appropriate place to ask the following questions: 1. Anyone please recommend open-source implementation of PoC (Push-to-talk Over Cellular) application server? Kamailio can do this? 2. Anyone please recommend open-source implementation of PoC related media server/replicator? Thierry Luo

3 2

Re: [SR-Users] authentication using auth_db
by arish haque 21 Apr '19

21 Apr '19

Hi David, This can solve the issue but depth(records) of that view would be too much. The architecture is like this: One kamailio fronting multiple asterisk. Each of asterisk boxes have their Specific users databases. Thus, I'm trying to create db connections inside request route or completely write my own authentication service if I get the user's Password. So, that would be great if I would be able to pass the db_url to auth_db module in request route. Let me know what you think. Thanks, Arish

2 1

Re: [SR-Users] authentication using auth_db
by arish haque 20 Apr '19

20 Apr '19

Hi David, Thanks for the reply. I checked out sqlops module, this will solve the problem only when I get the authentication password (secret) of the user inside request route. For this I tried checking kamailio pseudovars but haven't found one. Therefore, I am sticking with auth_db module.If this is possible by any other way do let me know. thanks, Arish

2 1

Kamailio, TLS, and LetsEncrypt
by Gaurav Bmotra 19 Apr '19

19 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i am trying to configure LetsEncrypt (tls) for kamailio sing following link https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ but unable to start kamailio ,i m getting these error ------->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Apr 19 11:23:57 ubuntufresh systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) Apr 19 11:23:58 ubuntufresh kamailio[5365]: Listening on Apr 19 11:23:58 ubuntufresh kamailio[5365]: Aliases: Apr 19 11:23:58 ubuntufresh kernel: kamailio[5378]: segfault at 0 ip 00007f8d6fdac646 sp 00007ffe8e2337f8 error 4 in libc-2.27.so [7f8d6fcfb000+1e7000] Apr 19 11:23:58 ubuntufresh kamailio[5365]: ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Control process exited, code=exited status=255 Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Failed with result 'exit-code'. Apr 19 11:23:58 ubuntufresh systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<------------------------------------ plz help -- *Regards:* Gaurav Kumar

3 5

auth by IP
by Yu Boot 19 Apr '19

19 Apr '19

Following code snippet from default kamailio.cfg never gives 403 if you smart enough to set "fromdomain" parameter on Asterisk to Kamailio's IP. How to fix it? I want password-based registration (which is OK now) and permit calls via Kamailio only from permitted IPs. # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; }

2 4

TLS implementation for kamailio
by Gaurav Bmotra 19 Apr '19

19 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i want to implement TLs for kamailio plz help -- *Regards:* Gaurav Kumar

3 2

Microsoft Direct Routing Assistance And Kamailio Integration
by Phillip Cooper 18 Apr '19

18 Apr '19

Hi All, I am looking for some assistance on setting up Microsoft Direct Routing with Kamailio as an SBC. I have a partial config working, I have rewrote the headers going to Direct routing and can place calls in to and out of Direct routing. I have problems with Call hang ups and choppy audio only on the audio stream from Direct Routing. 1) place a call in to direct routing, Hang the call up from the caller and Direct routing hangs up the call. But If you place a call in to Direct routing and hang up the call from direct routing the direct routing hangs up but the call is still active on the caller. IF you call from direct routing through kamailio to a callee and hang up from direct routing the call is hung up correctly. 2) on some calls made to or from direct routing, the audio stream coming from direct routing is sometimes choppy. Its always the audio from direct routing never the audio going to direct routing. My set up: Asterisk <-> Kamailio <-> Direct Routing SIP: Asterisk <-> Kamailio <-> direct routing RTP: Asterisk <-> Direct Routing Kamailio is rewriting the R-URI, TO, FROM, Contact headers going to Direct routing, as Microsoft need the sbc hostname to be in the contact header in a particular format. I am looking for help on the above or if someone would be kind enough to provide a working config that would be awesome! :) Many Thanks in advance for assistance. :) Phil --- This email has been checked for viruses by AVG. https://www.avg.com

1 0

authentication using auth_db module
by arish haque 18 Apr '19

18 Apr '19

Hi all, I'm trying to establish mysql database connection inside request route (at runtime) based on the type of user, as credentials are stored in multiple databases (may be different servers) for the users. Therefore, I am trying to find out that is there anyway to connect to mysql dynamically in request route or pass 'db_url' parameter dynamically to auth_db module. Hope this explains the requirement. Looking for suggestions. With Regards, Arish

2 1

How to append a body to send_reply() ?
by David Dean 18 Apr '19

18 Apr '19

Hello Kamailio gurus, I'm successfully using send_reply() with append_to_reply() to add some custom headers. Now I need to add some text into the reply body. How can I add a body to send_reply() ? Thx, Dave

2 7

After-parsing config file?
by David Villasmil Govea 18 Apr '19

18 Apr '19

Hello all, Kind of a weird question: is it possible to get a config file AFTER it’s been parsed? I.e: after it’s skipped all non-defined blocks, after substdef have been applied, etc. In the form of a normal config file Is this possible? -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

Use domain instead of IP in kamailio for register user
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i wan to to use domain instead of IP for registration in kamailio Plz help -- *Regards:* Gaurav Kumar

3 3

Does Kamailio provide any API for other program to create sip account
by Gaurav Bmotra 18 Apr '19

18 Apr '19

Hi i am Gaurav I am developing a sip VoIP app in Android, which want to provide the feature of allowing user to create sip account from client side. I use Kamailio as sip server. My thought is when user register to the the user account system, the user account system can communicate with Kamailio, create a sip account associated with the user and pass the sip account to the client, the client then can configure itself using the sip account. Does Kamailio provide any API or module for me to do this? Thanks! -- *Regards:* Gaurav Kumar

2 1

re-connect call in case of network issue
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 how can we implement re-connecting call instead of disconnect in case of network problem plz help thanks -- *Regards:* Gaurav Kumar

1 0

VRF routing with kamailio 5.2
by Jonathan Hunter 18 Apr '19

18 Apr '19

Hi All, Hope everyone in community is well. I was wondering if anyone could let me know if VRF is supported with kamailio 5.2 ? Essentially I am looking to put kamailio between two SBCs and bridge the networks together. So currently Server A 1.1.1.1 connects to Server B 2.2.2.2 and they pass calls fine, and I want to add kamailio in between without changing any IPs, so using VRF I have one network interface intA with 1.1.1.1 and the other intB with 2.2.2.2. What I want to do is listen on both interfaces, which is fine but bridge or force traffic between the two. Is this possible with kamailio? As doing force socket and listening on both 1.1.1.1 and 2.2.2.2 kamailio just loops and routes to itself, and force socket doesnt help. Do you have any suggestions? Would running two instances of kamailio on same server help or even introducing to separate kamailio instances and force routing between the two for requests. I am currently just testing with SIP OPTIONS and it is already proving tricky. Any help or comments appreciated. I appreciate this might not be the best solution however we have a lead time with providers to make changes which are proving quite slow and impractical so just trying to work a solution. Thanks Jon

2 1

integration kamailio with Freeswitch for video conferencing
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 and freeswitch 1.8 on debian 9 i want to configure kamailio with freeswitch for video conformance plz provide any reference guide plz help Thanks -- *Regards:* Gaurav Kumar

1 0

video chat with kamailio
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i m working on video chat app so how can i use kamailio for video chat OR i need to use other integration like freeSwitch with kamailio ? Plz help -- *Regards:* Gaurav Kumar

2 1

Support for TLS v1.3 (RFC 8446)
by g.Helio 18 Apr '19

18 Apr '19

2 1

Via shows wrong protocol of Kamailio
by Andrew Chen 18 Apr '19

18 Apr '19

Hi all, Got another anomaly in 4.1.5. I have this block of code which does t_relay_tcp: if ($var(isTpgw) == 1) { xlog("L_INFO","[$ci][$mi][CSeq $cs] isTpgw = $var(isTpgw)"); if (!t_relay_to_tcp()) { xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp....Error...\n"); t_on_failure("DISPATCHER_FAIL"); } else { xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp\n"); } } But when I looked at my sip capture tool, it showed the Via marked by the Kam to use UDP: INVITE sip:######@<some domain>:5060 SIP/2.0 Record-Route: <sip:206.81.191.30;r2=on;lr=on> Record-Route: <sip:206.81.191.30;transport=tcp;r2=on;lr=on> Call-ID: DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM To: "sip:6204857@sjomaintpsg50.fuzemeeting.com" <sip:6204857@sjomaintpsg50.fuzemeeting.com> From: "LifesizeSIP" <sip:LifesizeSIP@gs-3924.GSLAB.COM:5060>;tag=DL325a9e10c7;epid=062EE468 CSeq: 1 INVITE Max-Forwards: 69 Via: SIP/2.0/UDP 206.81.191.30;branch=z9hG4bKb16b.bd6a0791c798e3c859f8608993d584ae.1;i=5611 Via: SIP/2.0/TCP 10.101.199.105:5060;rport=53272;branch=z9hG4bK-fc8090ecf8-DL 206.81.191.30 is the Kam. I also log the activity as seen in the code with those xlog: Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM][42][CSeq 1] isTpgw = 1 Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM][42][CSeq 1] LifeSize Softphone 8.1.12 (Windows) Relaying [INVITE] from (10.101.19 9.105) to (206.81.191.72) via tcp So why did the Kamailio mark the Via header on the outgoing INVITE UDP? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 1

"481 Dialog/Transaction Does Not Exist" after sending SUBSCRIBE with Expires: 0
by David Dean 18 Apr '19

18 Apr '19

When a client sends a SUBSCRIBE with Expires: 0, Kamailio responds correctly with a "202 OK". But then Kamailio sends a NOTIFY to the same client with "Subscription-State: terminated;reason=timeout". The client rejects this NOTIFY with a "481 Dialog/Transaction Does Not Exist". Why does Kamailio send this NOTIFY, and is there any way to fix or disable it because it's always rejected by the client? The client is IMSDroid, and Kamailio is 5.2.2 with a default configuration. Packet 1 = Client sending SUBSCRIBE with Expires: 0 SUBSCRIBE sip:X.X.X.X:443;transport=tcp SIP/2.0Via: SIP/2.0/TCP 192.168.0.21:45336;branch=z9hG4bK-670576127From: <sip:username@domain.com>;tag=1372208921To: <sip:kamailio.domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6Contact: <sip:username@192.168.0.21:45336;transport=tcp>Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 1321654295 SUBSCRIBEExpires: 0Content-Length: 299Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERAccept: application/pidf+xmlEvent: presence Packet 2 = Kamailio responding with 202 OK SIP/2.0 202 OKVia: SIP/2.0/TCP 192.168.0.21:45336;branch=z9hG4bK-670576127From: <sip:username@domain.com>;tag=1372208921To: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 1321654295 SUBSCRIBEExpires: 0Contact: <sip:X.X.X.X:443;transport=tcp>Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 0 Packet 3 = Kamailio sending a NOTIFY NOTIFY sip:username@192.168.0.21:45336;transport=tcp SIP/2.0Via: SIP/2.0/TCP X.X.X.X:443;branch=z9hG4bKefe7.5f8ec435000000000000000000000000.0To: <sip:username@domain.com>;tag=1372208921From: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6CSeq: 3 NOTIFYCall-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0Content-Length: 0Max-Forwards: 70Event: presenceContact: <sip:X.X.X.X:443;transport=tcp>Subscription-State: terminated;reason=timeout Packet 4 = Client responding to the NOTIFY with "481 Dialog/Transaction Does Not Exist" SIP/2.0 481 Dialog/Transaction Does Not ExistVia: SIP/2.0/TCP X.X.X.X:443;branch=z9hG4bKefe7.5f8ec435000000000000000000000000.0From: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6To: <sip:username@domain.com>;tag=1372208921Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 3 NOTIFYContent-Length: 0 There are two entries in the Kamailio log: INFO: {1 1321654295 SUBSCRIBE e8cffa8f-948d-eb99-69ca-c9f18bdf11a0} presence [notify.c:1614]: send_notify_request(): NOTIFY sip:username@domain.com via sip:Y.Y.Y.Y:45336;transport=tcp on behalf of sip:kamailio.domain.com for event presence : e8cffa8f-948d-eb99-69ca-c9f18bdf11a0 ERROR: {2 3 NOTIFY e8cffa8f-948d-eb99-69ca-c9f18bdf11a0} presence [subscribe.c:497]: delete_subs(): Failed to delete subscription from memory [slot: 174 ev: sip:kamailio.domain.com pu: presence ci: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0 ft: 1372208921 tt: c4eb72dabf44e7588b972bd3b454ea11-0cd6] These are the presence settings in the Kamailio config file: #!ifdef WITH_PRESENCE# ----- presence params -----modparam("presence", "db_url", DBURL)modparam("presence", "send_fast_notify", 0) # ----- presence_xml params -----modparam("presence_xml", "db_url", DBURL)modparam("presence_xml", "force_active", 1)modparam("presence_xml", "force_dummy_presence", 0)#!endif Appreciate your help.

2 1

Rewrite private IP in SDP
by Bonjour Madame 18 Apr '19

18 Apr '19

Hello all, I am dealing with a no audio issue, phones are not even ringing. There is a private IP address in the "VIA" section of the message header of the INVITE: Via: SIP/2.0/UDP 192.168.1.25:65291 ;rport;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34 and in the contact too: Contact: <sip:5002@192.168.1.25:65291;ob> In the "status 100 Attempting to connect your call" from the server, I can see the public IP of the phone under received: Via: SIP/2.0/UDP 192.168.1.25:65291 ;rport=53990;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34;received=PHONE_PUBLIC_IP And in the "status 200 ok" from the server, I have: Via: SIP/2.0/UDP 192.168.1.25:65291 ;received=PHONE_PUBLIC_IP;rport=53990;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34 Record-Route: <sip:SERVER_PUBLIC_IP;lr=on;ftag=9284528256a54f8b81e692133a15684c> I don't know what steps to take to resolve this issue.

3 3

Record route dynamic IP
by eyas barhouk 17 Apr '19

17 Apr '19

hello dears i'm trying to build kamailio & asterisk with the following : * Kamailio will listen to All the interfaces : listen=udp:0.0.0.0:5050 listen=udp:0.0.0.0:5060 * it will add the record header as in the following part : if (!isflagset(FLT_RR)) record_route(); setflag(FLT_RR); when the sip message get from kamailio to asterisk the record route header will be like "<sip:0.0.0.0:5060;....etc.>" So is there any way to make the Via and record route headers has a specific IP instead of (0.0.0.0) i mean Something like "record_route("$Ri")". thanks & regards

2 1

How can i change kamailio SIP port form 5060 to any other
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i want to change kamailio SIP port 5060 to other port for security purpose plz help me -- *Regards:* Gaurav Kumar

2 3

TCP connection lifetime
by Aymeric Moizard 17 Apr '19

17 Apr '19

Hi Again, Here is an issue with TCP connection being kept for more: Yesterday, I have discovered that a User-Agent (<Avaya IP Phone 1120E (SIP1120e.04.04.30.00)> tried to register a lot. It was sending REGISTER over new established TCP socket *every 2 seconds*. All the REGISTER was rejected with 401. (may be the device was misconfigured? or not receiving any of my answer? I can't tell) NOTE: You can see the expires header was very large: 86400, ie: 24 hours... I was checking the TCP/TLS connections on my server and discovered more than 1000 TCP established connection to that user/ip, and thus, I have tried to understand what happened. Checking the logs, I received 4855 REGISTER from this device from "Mar 25 03:47:09" to "Mar 25 07:56:13" which is a rate of approx one new TCP connection every 2.5 seconds... Today, I decided to check it again around 11am. jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1857", "tcp:con_timeout = 35927", "tcp:connect_failed = 25", "tcp:connect_success = 2", "tcp:current_opened_connections = 2291", "tcp:current_write_queue_size = 0", "tcp:established = 80778", "tcp:local_reject = 0", "tcp:passive_open = 80776", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 7305 } There was still A LOT of established connections. And the connections have been established more than 24 hours ago. At 11H16: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1161 At 11H22: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1018 At 11H35: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 655 At 13H $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 0 So the established connections are all gone now. Between 11h16 and 11H35, I was seeing the server regularly sending [FIN, ACK] over each TCP established connection, with retransmissions for all of them. (no incoming trafic) I do not have numbers/capture/stats, but I think that kamailio was already closing some connection yesterday. I don't know when kamailio started to try closing those connections. I'm now back with this status: At 13pm: jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1896", "tcp:con_timeout = 38042", "tcp:connect_failed = 26", "tcp:connect_success = 2", "tcp:current_opened_connections = 939", "tcp:current_write_queue_size = 0", "tcp:established = 81950", "tcp:local_reject = 0", "tcp:passive_open = 81948", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 12734 } With around 155 registration entries using TCP and TLS in my location database. As you can see, tcp:current_opened_connections = 939 is still pretty high compared to my currently registred users. I have "modparam("registrar", "max_expires", 86400)", because I'm keeping contact entries (even with TCP connection down) for push notifications. I have "tcp_connection_lifetime=3600" configured. Question 1 With "tcp_connection_lifetime=3600", I would expect kamailio to close the established connection after 3600 seconds without traffic. It is pretty obvious that no data has been exchanged over the 4855 established connection during a day. Despite the issue with the Avaya phones is solved automatically after a day, I guess similar stuff or happening, at a different rate, for other users as well. (because current_opened_connections is way higher than registred TCP/TLS users) Question 2 I can list TLS connection with "kamctl rpc tls.list" Can I get a similar list for TCP? (lsof returns a lot of duplicates...) Tks Aymeric -- Antisip - http://www.antisip.com

2 4

Route header
by Duarte Rocha 17 Apr '19

17 Apr '19

Greetings, Is there a difference for Kamailio between these 2 situations. 1 - Method with 2 Route Headers, each one with one address and one uri 2 - Method with 1 Route Header that includes 2 Route URI Best Regards, Duarte Rocha

1 0

allow calls only from registered users
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i wan to implement user based calling on kamailio , that only can call how registered with user plz help -- *Regards:* Gaurav Kumar

2 1

Send 200 OK response with body to INVITE requests
by David Dean 16 Apr '19

16 Apr '19

Hi all, I'm building an automated load testing system for an rtp media server with TURN that uses Kamailio for SIP. When a client sends an INVITE to Kamailio, I need Kamailio to respond with "200 OK" and the SDP generated by rtpengine. I've enabled auto-answer in the client, so as soon as it receives the response from Kamailio it will start sending RTP packets. Right now I'm using send_reply("200", "OK"); which is sending the right 200 response back to the client, but it doesn't include any body. How can I get Kamailio to send a "200 OK" and add the rtpengine_answer into the body? Appreciate any help, I've been struggling with this for three days. Cheers. rtpengine_offer("SIP-source-address ICE=force-relay RTP");send_reply("200", "OK");

4 7

Getting a new RTP Port on reINVITE during answered call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure (fake IPs) 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ //listen=udp:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" /**************/ loadmodule "usrloc.so" /**************/ loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

new RTP Port after INVITE during call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ //listen=udp:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" /**************/ loadmodule "usrloc.so" /**************/ loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN //route(PSTN); # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN xlog("L_INFO","Entering PSTN"); # route to PSTN dialed numbers starting with '+' or '0000' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) { xlog("L_INFO","$rU is not outbound call to PSTN"); return; } if (!starts_with("$si","10.19.139")) { xlog("L_INFO","$rU is not comming from any Media GW"); return; } # only local users allowed to call if(from_uri!=myself && !ds_is_from_list("1") ) { xlog("L_INFO", "from_uri $fu != myself, so this is Not Allowed!!"); sl_send_reply("403", "Not Allowed"); exit; } xlog("L_INFO","$si request"); xlog("L_INFO","$rU is an outbound call to PSTN"); $ru = "sip:" + $rU + "@62.28.173.10"; // + $sel(cfg_get.pstn.gw_ip); route(RELAY); /* if(!ds_select_dst("20", "4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); */ exit; #!endif xlog("L_INFO"," NO PSTN SET!!"); return; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Dispatch requests route[DISPATCH_TO_PSTN] { # round robin dispatching on gateways group '20' - PSTN if(!ds_select_dst("20", "10")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

single Contact per AOR dilema
by Sergiu Pojoga 16 Apr '19

16 Apr '19

Hi ppl, Say we want to allow 1 contact per AOR. modparam("registrar", "max_contacts", 1) Say some UA looses connection before de-registering, then comes back shortly after (before Contact expires) and tries to register with a different Contact, but gets rejected with error "test_max_contacts(): too many contacts for AOR". What's the approach to handle this situation? I'd like the registrar to accept and save that AoR based on the latest registration. Similar to how Asterisk's single-contact stack `chan_sip` works - whoever was the latest to register successfully gets to be the winner. Thanks, --Sergiu

2 2

dlg_onroute(): failed to update dialog lifetime
by Olli Attila 15 Apr '19

15 Apr '19

Hello, I just activated snmpstats module on Kamailio 5.0.7 and added the following line to route(INITIAL) at the bottom of the route like this (For noc group to monitor Kamailio dialogs): # Per SIP request initial checks route[INITIAL] { ... dlg_manage(); } Now I started to gets this error few times in an hour: /usr/sbin/kamailio[22194]: CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f798ca51b98 tl->next=(nil) tl->prev=(nil) Apr 15 09:54:05 /usr/sbin/kamailio[22194]: ERROR: dialog [dlg_handlers.c:1439]: dlg_onroute(): failed to update dialog lifetime I see no call failures because of this but still interested what is happening here? --Olli Attila

1 0

one way Audio issue
by Gaurav Bmotra 12 Apr '19

12 Apr '19

i have installed kamailio 5.1.x on Ubuntu server 18.4 issue is i am able to connect the call but there is one way audio not both way as i have install rtpproxy also with following config /etc/default/rtpproxy EXTRA_OPTS="-F -s udp:127.0.0.1:7722 -l PUBLIC_IP -d DBUG:LOG_LOCAL0" /etc/kamailio/kamailio.cnf modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") after restart kamailio i got this logs dev /usr/sbin/kamailio[6926]: INFO: rtpproxy [rtpproxy.c:1559]: rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled during calling i got this logs Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): dr Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error whi Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db_request(): failed to insert in Apr 12 16:05:43 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13475]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction ans Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): drive Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db Thanks Gaurav *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] [sr-dev] Audio missing during call
by Mikko Lehto 12 Apr '19

12 Apr '19

Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > HI i have install Kamailio 5.1.2 on Ubuntu 18.4 > there is a audio issue , when we call call from one extension to another ,, > in bigning call goes good , after 9-10 sec it miss audio for one side (one > side can hear other side can't hear ) Hi, mailing list sr-dev is for Kamailio development related topics, please subscribe to sr-users for more general queries. Before posting, make sure you have described in detail your environment and also troubleshooting steps you have done so far: - what modifications you made to default configuration - what happens during SIP transactions in fail scenario - are you using RTP relay or not (rtpproxy/rtpengine) - what have you observed from packet captures Good luck -- Mikko

2 1

command out of sync with db_cluster and db_mysql
by Simone Lazzaris 12 Apr '19

12 Apr '19

Hi everybody; I'm trying to use db_cluster to connect my kamailio installation to a mariadb galera cluster, and it's driving me crazy. In my lab it works without issues, but when I try the setup in the production node, I get some strange errors: Apr 11 14:21:31 ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): driver error on query: Commands out of sync; you can't run this command now (2014) Apr 11 14:21:31 ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query This is my configuration: modparam("db_cluster", "connection","con1=>mysql://openser:PASSWD@X.X.X.30/ openser") modparam("db_cluster", "connection","con2=>mysql://openser:PASSWD@X.X.X.31/ openser") modparam("db_cluster", "connection","con3=>mysql://openser:PASSWD@X.X.X.32/ openser") modparam("db_cluster", "cluster", "cls1=>con1=9r8r;con2=9r8r;con3=9r8r") #!define DBURL "cluster://cls1" #!define SQLOPSURL "database=>cluster://cls1" modparam("dispatcher", "db_url", DBURL) modparam("acc", "db_url", DBURL) modparam("usrloc", "db_url", DBURL) modparam("auth_db", "db_url", DBURL) modparam("permissions", "db_url", DBURL) modparam("alias_db", "db_url", DBURL) modparam( "sqlops", "sqlcon", SQLOPSURL ) I'm using version 5.1.4 of kamailio, freshly compiled from source on a debian 9. Any ideas? *Simone Lazzaris* *Qcom S.p.A.* simone.lazzaris(a)qcom.it[1] | www.qcom.it[2] * LinkedIn[3]* | *Facebook[4]* [5] -------- [1] mailto:simone.lazzaris@qcom.it [2] https://www.qcom.it [3] https://www.linkedin.com/company/qcom-spa [4] http://www.facebook.com/qcomspa [5] https://www.qcom.it/includes/email-banner.gif

2 2

kamailio, asterisk and BLF seems its only using json?
by PICCORO McKAY Lenz 12 Apr '19

12 Apr '19

hi, i try to follow the how to in the wiki https://kb.asipto.com/kamailio:presence:k43-blf and configure my two only phones.. but for any states only one led its blinking, the busy state.. no other state work, (if i connect directly to asterisk all work perfeclty, i using asterisk 13 with kamailio 4.4.4) also read the archived https://lists.kamailio.org/pipermail/sr-users/2013-May/078234.html but no luck wht happened with that how to? it's different for asterisk if are integrated? do i need a pjsip asterisk event the oldes chan? see that in presense module parameters, the server address said that value of Contact header filed for 200 OK replies to SUBSCRIBE and PUBLISH and in NOTIFY messages umm if i have integrated and working asterisk (relatime mode) with kamailio, that address muswt be asterisk? taking in cosideration that i have a route rule said that send auth to asterisk and trust any incomming from it? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

1 0

having issue updating $td
by Andrew Chen 11 Apr '19

11 Apr '19

Hey guys, Question. I have this block of code at the beginning of route block: if(is_ip($td)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] IP detected in To domain. Resolving. \n"); exec_avp("host $td | awk '{print substr($$5, 1, length($$5)-1)}'", "$avp(s:newtd)"); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new newtd = $avp(s:newtd)"); $td = $avp(s:newtd); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new td = $td"); } but $td never updates. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] IP detected in To domain. Resolving. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new newtd = sjomaintpsg50.fuzemeeting.com Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new td = 206.81.191.30 Is there a reason why? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 3

Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

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Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

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Releasing Kamailio v5.1.8
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, I am considering to release v5.1.8, an minor version to be built from branch 5.1, on Thursday, this week (April 11, 2019). As usual, check that at least the patches for issues you reported were backported in order to be sure they were not missed during the typical backporting done by developers so far. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

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Using AWS boto3 with KEMI Python routing scripts
by Michael Loughrey 10 Apr '19

10 Apr '19

Hi All, looking for some advice regarding the proper way to initialise an AWS API boto3 client object for send SNS messages within a KEMI Python 2.7 routing script. I'm sure process forking has a major impact on how this works - from kamailio.cfg fork=yes children=4 - but would greatly appreciate some guidance. I have tried various methods to allocate an client object self.sns_client = boto3.client('sns', region_name=MY_AWS_REGION) 1. within __init__(), only once from module initialisation 2. on demand within a function call by any client code e.g. called from ksr_route_request(), each creating it's own client object and so far all have resulted in intermittent crashes within botocore/client.py ultimately crashing thus : Core was generated by `/usr/local/sbin/kamailio -P /usr/local/kamailio/run/kamailio.pid -f /usr/local/'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 (gdb) bt #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #1 0x00007f8f9734f82e in X509_VERIFY_PARAM_free () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #2 0x00007f8f97642f5c in SSL_free () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #3 0x00007f8f86b09c64 in PySSL_dealloc.lto_priv.5 () at ./Modules/_ssl.c:1598 #4 0x00007f8f99cbb007 in insertdict_by_entry (mp=0x7f8f841f1e88, key='_sslobj', hash=<optimized out>, ep=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:519 #5 0x00007f8f99cbe2cf in insertdict (value=None, hash=1051385741686792393, key='_sslobj', mp=0x7f8f841f1e88) at ../Objects/dictobject.c:556 #6 dict_set_item_by_hash_or_entry (value=None, ep=0x0, hash=1051385741686792393, key='_sslobj', op={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/dictobject.c:795 #7 PyDict_SetItem (op=<optimized out>, key=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:848 #8 0x00007f8f99becec1 in _PyObject_GenericSetAttrWithDict (obj=<optimized out>, name='_sslobj', value=None, dict={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/object.c:1529 #9 0x00007f8f99bed437 in PyObject_SetAttr ( v=<SSLSocket(server_hostname=u'sns.us-east-1.amazonaws.com', _connected=True, _context=<SSLContext at remote 0x7f8f84232398>, server_side=False, _makefile_refs=0, _closed=False, _sslobj=None, do_handshake_on_connect=True, suppress_ragged_eofs=True) at remote 0x7f8f841f62a8>, name=<optimized out>, value=None) at ../Objects/object.c:1247 Here's my setup host : Debian9/stretch python --version Python 2.7.13 kamailio -v version: kamailio 5.2.2 (x86_64/linux) 67f967-dirty flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, MEM_JOIN_FREE ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 67f967 -dirty compiled on 23:44:31 Apr 8 2019 with gcc 6.3.0 Cheers Mike

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Rtpengine - no RTP packets going out
by Istvan Mogyorosi 10 Apr '19

10 Apr '19

Dear all, This is my first post after reading a lot in this mailing-list. I'm trying to use Kamailio 5.1 with the dispatcher module and rtpengine acting as SIP + RTP proxy. I have 6 asterisk servers in a private subnet that should talk with the peer via a single IP like this: Asterisk 1..n|---> | GW.PRIVATE.IP -o- GW.PUBLIC.IP |----> PEER.SIP.TRUNK I'm on Centos 7, with firewalld configured, iptables module is loaded and the rule is well defined. Packet forwarding is also enabled. Chain rtpengine (1 references) target prot opt source destination RTPENGINE udp -- anywhere anywhere RTPENGINE id:40 My call flow seems to be fine, Kamailio/rtpengine private IP is the outboundproxy parameter of Asterisk instances. My problem is that RTP packets are not present on the public interface, the rtpengine final log showing the 2 sessions, but I'm not sure this is what I want or simply the firewall does not let it out ? (To be more precise PEER.SIP.TRUNK is the trunk for SIP traffic, I have multiple IP addresses for media to connect to, reinvites are allowed) Closing call due to timeout Final packet stats: --- Tag 'as6d12caea', created 1:00 ago for branch '', in dialogue with 'as541b1e61' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PRIVATE.IP:10000 <> 192.168.30.13:11152, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PRIVATE.IP:10001 <> 192.168.30.13:11153 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts --- Tag 'as541b1e61', created 1:00 ago for branch '', in dialogue with 'as6d12caea' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PUBLIC.IP:10000 <> PEER.SIP.TRUNK:28216, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PUBLIC.IP:10001 <> PEER.SIP.TRUNK:28217 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts Best regards, Istvan

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Kamailio and CDR
by Yu Boot 10 Apr '19

10 Apr '19

Is there any other way to retrieve CDRs from Kami other than dialog/acc modules and writing CDRs to syslog only? Thanks.

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Help request in regards to Kamailio
by Anas ZAHHAF 10 Apr '19

10 Apr '19

Dear, I am writing to ask for help, I found your Email after I've subscribed to Kamailio.org. I am a beginner in Linux environment ( I worked on Windows for 20 years), it would be very helpful if you can help me installing Kamailio on a computer running Debian and how to use the load balancing module so that I can split the charge between 2 asterisk servers, by sharing any useful tutorial or document or even by Teamviewer if this doesn't disturb you. Please help me, all the tutorials I find on internet aren't made for beginners. I will be awaiting your reply and excuse me again if I disturb you. Best regards, -- *Anas ZAHHAF* *105, rue des Moines75017 PARIS* *+33(0)7.66.31.31.85* azahhaf(a)zaion.ai <niluse(a)zaion.ai> | www.zaion.ai ᐧ

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change failure status if reason is q.850 cause=31
by Techinnovation 10 Apr '19

10 Apr '19

Hello, How can I change the status where Reason: q.850;cause=31. ? This code change status for all where we receive status 480 to 404. Is it possible to this code also to add second condition - if reason is like "q.850;cause=31" failure_route[MANAGE_FAILURE] { ... if (t_check_status("480")) { t_reply("404","Unallocated number"); exit; } Need to modify this massage. SIP/2.0 480 Temporarily Not Available. Allow: UPDATE,REFER,INFO. Call-ID: 6b11b153405a5103330697b774b97f5a@192.168.2.2:5060. Contact: <sip:192.168.3.3:5060>. CSeq: 102 INVITE. From: "380668375820" <sip:380668375820@192.168.2.2>;tag=as445242c0. Reason: q.850;cause=31. Server: Cirpack/v4.76 (gw_sip). To: <sip:9052946032@192.168.3.3:5060>;tag=00-07929-6e4a3079-75447e2a5. Content-Length: 0. Thanks. BR, Aleksandr

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Question regarding retcode
by Andrew Chen 09 Apr '19

09 Apr '19

Hey all, Question regarding the type of retcode. I ran a curl command and $retcode returned 200. I check against that variable if it's 200 or not. I tried to check 200 as a string and integer but never matches so I ran these lines of code: xlog("L_INFO", "[$ci][$mi][CSeq $cs] Validating meeting ($rU). Result: $var(result) Return code: $retcode \n"); if(is_int($retcode)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is an integer \n"); } if(typeof("$retcode", "str")) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is a string \n"); } None of these xlog outputed. How do I check for $retcode in if or switch statement? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

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Centos 7 opensuse repo issues for 5.2
by Jeffrey Gray 09 Apr '19

09 Apr '19

Good afternoon, We are unable to get kamailio 5.2 to install using the recommended opensuse repos for Centos 7. Is there any way to resolve the following errors: Thanks very much for your assistance! using the following repo setup in /etc/yum.repos.d/ [home_kamailio_v5.2.x-rpms] name=RPM Packages for Kamailio v5.2.x (CentOS_7) type=rpm-md baseurl=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rp… gpgcheck=1 gpgkey=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpm… enabled=1 Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: kamailio-5.2.2-3.el7.centos.x8 FAILED ================- ] 2.3 MB/s | 1.7 MB 00:00:01 ETA http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.2.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try. If we run the suggested clean metadata command, then the error changes to: sudo yum -y install kamailio Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirrors.advancedhosters.com * epel: mirror.cogentco.com * extras: mirrors.advancedhosters.com * updates: mirrors.advancedhosters.com Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-5.2.2-3.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try.

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