sr-users June 2018

sr-users@lists.kamailio.org

88 participants
119 discussions

Problem with TLS
by Arik Halperin 12 Jun '18

12 Jun '18

Hello, I’m using TLS After receiving 200OK from kamailio: r2voip.clear2voipdialer I/(NativeSdk_2_0) 1528174138320 PJSIP: (NativeSdk_2_0) 1528174138320 PJSIP:2018-05 07:48:58.319 pjsua_core.c RX 2203 bytes Response msg 200/INVITE/cseq=8107 (rdata0x7a2c56fb38) from TLS 70.36.25.65:443: SIP/2.0 200 OK Via: SIP/2.0/TLS 10.134.232.109:44097;received=109.253.173.146;rport=31373;branch=z9hG4bKPj4MV5llP9SW5ufk-OcFB-Qh78PmIQFrRk;alias Record-Route: <sips:10.168.10.227:5099;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> Record-Route: <sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> From: "number" <sips:972523391991@kamprod.telemessage.com<mailto:972523391991@kamprod.telemessage.com>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO To: <sips:1111111@kamprod.telemessage.com<mailto:1111111@kamprod.telemessage.com>>;tag=64H63g861ajHj Call-ID: Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2 CSeq: 8107 INVITE Contact: <sip:1111111@10.168.10.200:5080;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.6.20+git~20180123T214909Z~987c9b9a2a~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: ti PJSIP responds with: Secure dialog requires SIPS scheme in Contact and Record-Route headers, ending the session What is the reason for this? How can I fix this issue? Thanks, Arik Halperin

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Disable basic authentication?
by Skip Morse 11 Jun '18

11 Jun '18

Hi All, I couldn't get a clear answer from the documentation. Based on the description of www_authenticate and proxy_authenticate: " The function verifies credentials according to RFC2617. If the credentials are verified successfully then the function will succeed and mark the credentials as authorized (marked credentials can be later used by some other functions). If the function was unable to verify the credentials for some reason then it will fail and the script should call www_challenge which will challenge the user again." The RFC outlines basic and digest auth. Is there a way to disable the ability for a UA to use basic auth? Or maybe these is disabled already? I'm likely understanding this wrong, it seems to me that it's allowed for a UA to auth with basic even if the challenge is for digest. Any clarification would be helpful. Thanks! -Skip

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SIP/2.0 603 Failed to get local SDP for sip to webrtc (SIPML5)call
by vinod mn 11 Jun '18

11 Jun '18

Hi i configured sipml5 and kamailio , sipml5 clients registers to kamailio when i make a call to kamailio call lands on the sipml5 client but i am getting this error *"SIP/2.0 603 Failed to get local SDP" *on chrome/firefox/opera webbrowsers. kamilio configuration as per http://www.kamailio.org/docs/ modules/4.4.x/modules/websocket.html#idp53411716 -- Thanks and regards Vinod.M.N

2 1

sip digest authentication
by eyas barhouk 11 Jun '18

11 Jun '18

Hello dears I’m trying to setup Kamailio with sip digest authentication scheme by using the following configuration from s-cscf side : modparam("ims_auth", "registration_default_algorithm", "HSS-Selected") and the result was as following : 1. when the register message passed form the s-cscf to the hss and in the diameter trace I found the “SIP-AUTHENTICATION-SCHEME” in the MAR request as “Digest-MD5” 1. After that the HSS send MAA message with “DIAMETER-ERROR-AUTH-SCHEME-NOT-SUPPORTED” 2. S-cscf replay with the message “Forbidden – privet identity not found (Authorization : username)” So based on that could any one tell where was my mistake or how can I setup Kamailio with SIP-DIGEST scheme Thanks

2 4

JSON RPC Error: LUMP_RPL_BODY already added!
by Asgaroth 11 Jun '18

11 Jun '18

Hi All, We are trying to query the json rpc server on our registrar for the contact of an aor, but we are seeing the following error(s) being displayed in the log when we query the location table: ERROR: <core> [core/data_lump_rpl.c:83]: add_lump_rpl2(): LUMP_RPL_BODY already added! ERROR: xhttp [xhttp_mod.c:410]: xhttp_send_reply(): Error while adding reply lump We are getting the aor details back in response from the rpc call, but the error's in the log are a little concerning and I cannot see where we are doing something wrong. As an example, the following curl request returns the correct response, but the above error is displayed in the log, Am I doing something wrong here, I cannot see what the issue is: curl --header 'Content-Type: application/json' --data-binary '{ "jsonrpc": "2.0", "method": "ul.lookup", "params": [ "location", "user(a)example.com" ], "id": 3 }' http://registrar.example.com/rpc A secondary question is, is there an upper limit on the size of the "id" parameter? This kamailio instance is v5.0.6. Any thoughts as to what this error means? Thanks

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(N)DB_FIREBASE module?
by Pan Christensen 11 Jun '18

11 Jun '18

Greetings all. We are currently developing web and mobile apps using firebase: https://firebase.google.com/products/realtime-database/ . We have been discussing the possibility of Kamailio querying this database. I assume that it will be possible to query it using HTTP and JSON modules, but it would probably be easier to use the DB API. Any plans to develop such a module? With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobile: 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 10 Jun '18

10 Jun '18

I just did a test where I disabled VP8 in Kamailio using SDPOPS and I now get Video on the softphone however, I lost two-way Audio. Kamailio seems to be doing something with the codecs but still can't put my finger on it. The WebRTC Client, who is the caller, needs VP8 for Video, and apparently Audio. I'm not sure if I getting closer or just moving the problem around. Thanks All! From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Saturday, June 9, 2018 12:47 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio >>>Email originates from a non-MITRE system. Use caution.<<< I am desperately trying to resolve a big issue I have when using Kamailio 5.2.0 and Asterisk 15.3. As soon as I go through Kamailio, I get no Video on either side of the call; I do get two-way audio and the call stays connected. If I simply go right through Asterisk, I also get two-way Video. I am having a tough time determining why Kamailio is messing with the Video portion of the call. I have included a full pcap file. As a side note when comparing pcap traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from Asterisk and Softphone are exactly the same. Thank you, -Steve

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No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 09 Jun '18

09 Jun '18

I am desperately trying to resolve a big issue I have when using Kamailio 5.2.0 and Asterisk 15.3. As soon as I go through Kamailio, I get no Video on either side of the call; I do get two-way audio and the call stays connected. If I simply go right through Asterisk, I also get two-way Video. I am having a tough time determining why Kamailio is messing with the Video portion of the call. I have included a full pcap file. As a side note when comparing pcap traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from Asterisk and Softphone are exactly the same. Thank you, -Steve

1 0

Combining lookup_branches() and t_(next|load)_contacts()
by Nathan Ward 09 Jun '18

09 Jun '18

Hi, About a year ago I posted this, but haven’t had much chance to follow up on it: https://lists.kamailio.org/pipermail/sr-users/2017-March/096337.html I’ve been doing some work on this issue (we worked around it until now), and have reduced the config right down to see where things are going wonky. The scenario I have, is inbound calls (to let’s say “A@domain") which I have aliased to two or more different numbers (let’s say “B@domain” and “C@domain”). Both B and C are registering, but are not always registered. I resolve the aliases with alias_db_lookup() (in my sample below, I use seturi() and append_branch() to avoid any confusion about where things may be going weird), then look them up in the location table with lookup_branches(). I then need to do t_load_contacts() and t_next_contacts(), as in some call scenarios I add a serial fork before or after the main call (pre-call announcements etc.). Then, I call t_relay(). This works great most of the time - however, if for example B is registered, but C is not registered, lookup_branches() notes that B is "Not found in usrloc”, and then t_load_contacts() seems to not realise that this isn’t usable, and it attempts to do a DNS lookup for the domain and so on, which I don’t really want it to be doing. Should lookup_branches() be removing a branch it can’t resolve? I would expect it to behave like that, anyway. Can I configure it to do that, or do I need to manually loop through them or something to make sure they resolved, and remove the ones that didn’t? I’ve tried swapping it around, so I call lookup_branches() after t_next_contacts() but the behaviour is the same. I imagine I can’t be the only person trying to do this - how have others made it work? Config sample: seturi('sip:a@domain'); append_branch('sip:b@domain'); xlog("L_DBG", "1 ds: [$ds]\n"); lookup_branches('location'); xlog("L_DBG", "2 ds: [$ds]\n"); t_load_contacts(); xlog("L_DBG", "3 ds: [$ds]\n"); t_next_contacts(); xlog("L_DBG", "4 ds: [$ds]\n"); -- Nathan Ward

1 1

Peculiar Kamailio Asterisk behavior on outbound calls
by Wilkins, Steve 09 Jun '18

09 Jun '18

Hi All, Issue: when a Call is made through Kamailio and Asterisk. Asterisk uses incorrect Video RTP Payload Type when sending Video packets. I have a situation where I make a call from Device A to Device B and Device A is Registered in Kamailio. When Device A Calls Device B, Kamailio sends an 'INVITE' to Asterisk, Asterisk then 'INVITES' Device B. I get two-way Audio, the call stays connected, however, when Video packets are sent to Device B, the RTP Payload Type is incorrect. The port is correct, but just not the Payload Type. Here is where I think Kamailio is involved. In the first Invite from Kamailio to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, Device B wants to use '115 H264' and when Asterisk sends out Video packets, it is using '100' instead of '115', and of course I have no Video. I don't know if this is just a coincidence but it sure seems like that is where the issue may lie. Has anyone ever seen this behavior? The Asterisk teams does not think it's an Asterisk issue. Thank you, -Steve

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sr-users June 2018