sr-users May 2019

sr-users@lists.kamailio.org

84 participants
97 discussions

by Gaurav Bmotra

HI i am using kamailio 5.1.2 With Ubuntu 18.04.2 LTS and using Freeswitch as Media server and kamailio as Register server i wan to implement load balance on this plz help thank you Gaurav -- *Regards:* Gaurav Kumar

5 years, 1 month

usrloc extra attributes and db_mode=0

by Patrick Wakano

Hello list, Hope you all doing well! I am trying to use the extra attributes (xavp_contact) of the usrloc module to save some additional info about the user. I am setting the value before the save() and doing a call to registered() (not lookup()) before trying to access these extra attributes. This works fine with db_mode=3 but does not work in case of db_mode=0. Does anyone knows if it should also work with mode 0? I was expecting it to work but a call to kamcmd ul.dump show nothing related..... I don't want to use the DB for the location purposes because I prefer to have multiple servers using the dmq_usrloc (that works like a charm!) and it conflicts with DB persistence of the location table (when the DMQ replication happens, all servers (sharing the same DB) try to save the same user info in the location table) Thank you, Kind regards, Patrick Wakano

5 years, 1 month

Statistics help

by Duarte Rocha

Greetings, I'm doing some monitoring of my Kamailio in order to quickly identify system failures and overloads. In order to do that i have scripts that read the output of "kamcmd sl.stats" and identify the rising of "500_replies", "5xx_replies" and "6xx_replies". However, i have rate limiting on my Kamailio, which generates "503" replies when limit is reached, which generates an unnecessary alarm. Is there a way to get more specific stats about the repleis generated by Kamailio? If that isn't possible, will i be safe only monitoring "500" and "6xx" replies? In my experience, when the server has reached a processing or memory limit, the reply is always 500, but i'm not really sure about this. Thank you for your help. Best Regards, Duarte Rocha

5 years, 1 month

Push proxy and load balancer for asterisk?

by Gerrit Jacobsen

Good morning, I want to realise a push notification proxy to signal incoming calls from asterisk to softphones. Is Kamailio the preferred solution for this? I looked also at Flexisip and Asterisk scripts and was told by a seasoned professional that Kamailio is the preferable solution as its scalable. True? I would be glad if someone could share his experiences. Also can one Kamailio instance be used as an asterisk load balancer and also push notification proxy ? I have been a heavy openser user a long time ago. I used first openser as my core routing engine but then switched to yate. Please excuse my rather broad questions. From my experience with Openser the devil is in the detail. So any hints on where this setup may cause problems would be great. Thanks a lot for your input. Gerry

5 years, 1 month

Kamailio Lord Balance

by Gaurav Bmotra

hi i m using kamailio 5.1 with Ubuntu 18.04.2 LTS and freeswitch as Media server -- *Regards:* Gaurav Kumar

5 years, 1 month

RFC: split out old sr api from app_lua

by Daniel-Constantin Mierla

Hello, I am considering to split the app_lua module in order to move the older style API available under Lua 'sr' module in a dedicated module (like app_lua_sr). Later this module can be obsoleted and removed, once we know that everything there is covered by KEMI KSR Lua module. In this way, we make it more clear that KSR is the preferred API to use as well have the app_lua smaller, for those using only KSR and not needing 'sr' API. Therefore I am starting the discussion to see if anyone has something against or other suggestions. I cc-ed sr-dev to make developers aware, but I think the best place for discussion is sr-users, being related to how Kamailio is used, therefore reply only to sr-users. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

5 years, 1 month

RTPEngine / Transcoding question

by Carsten Bock

Hi, I want to implement selective transcoding, e.g. avoiding Transcoding between a HD codec (G722) and a non HD Codec (G711). In case the caller does offer HD codecs, this is fine, e.g. if(sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_offer("codec-transcode=OPUS codec-transcode=G722 codec-transcode=PCMA"); } else { rtpengine_manage("codec-transcode=PCMA"); } Now I have the issue, if the callee only sends me G711, I don't want to offer G722 or Opus to the caller. However, rtpengine_answer() does not seem to accept "codec-strip=G722 codec-strip=OPUS", e.g.: onreply_route() { if(!sdp_with_codecs_by_name("G722,OPUS")) { rtpengine_ manage("codec-strip=OPUS codec-strip=G722"); } } As a result, the SDP always contains the transcoding options, e.g. G722 and OPUS. I always end up in transcoding G722 to G711, which is meaningless in that case. Any ideas on how to solve that? (above examples very simplified) Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

5 years, 1 month

RTPEngine: rtpengine_manage(), Subsequent requests, Transcoding

by Carsten Bock

Hi, quick question: If understood it correctly in the past (and it worked for me quite well this way), I can simply call rtpengine_manage() for subsequent requests without adding any additional information (e.g. used interfaces, RTP/SRTP translation or anything like that), as this information is stored in the internal hash table of the RTPEngine module during call setup and initial offer/answer. However, this does not seem to apply to the transcoding options. When I just call "rtpengine_manage()" for a Re-INVITE of a transcoded call (in my case AMR-WB to G722), the transcoding options are no longer applied. The forwarded offer does not get the transcoded codecs added (e.g. in my case G722 is not added to the list of AMR/AMR-WB from the phone). Did I misunderstand something or is it a bug? My RTPEngine is still on 7.2, but I will double check with more recent versions today or tomorrow. However, I also found no commit messages relating to this. Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

5 years, 1 month

Call Routing at IMS

by Iman Mohammadi

Hi Everyone When P-CSCF receives INVITE from client it sends INVITE to S-CSCF then S-CSCF without any look up(checking the user is registered or not)sends INVITE to I-CSCF then I-CSCF sends LIR and receives LIA and sends INVITE to same S-CSCF (in case of one S-CSCF exists).My Question is here why without any look up S-CSCF sends INVITE to I-CSCF??this procedure affects system performance and increases call set up time

5 years, 1 month

WebRTC bad message (offset: 22)

by Ilie Soltanici

Hello, Can someone help me to discard those messages from kamailio log files? ERROR: <core> [core/parser/parse_fline.c:262]: parse_first_line(): parse_first_line: bad message (offset: 22) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TLS 212.2.161.34:60477#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: VeiXbJClWneIrIyYgM5v3XKYHLA=#015#012Server: WebRTC GW#015#012Content-Length: 0#015#012#015#012> As I understand this is just an HTTP response to the first request, and shouldn't go through config file execution, but I don't understand why it is getting through it. Thank you

5 years, 1 month

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sr-users May 2019