sr-users February 2018

sr-users@lists.kamailio.org

90 participants
109 discussions

PUBLISH and ETag
by Cyrille Demaret 16 Feb '18

16 Feb '18

Hi, I'm using Kamailio with presence enabled and Asterisk PJSIP and outbound-publish. My problem is happening when I place 2 consecutive calls from Asterisk : When I make a first call Asterisk sent the following: PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e69723a610 82 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Event: dialog Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early.. Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e6972 3a61082;received=192.168.100.37 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-ff39 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19863.16.0 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 When the call is done, Asterisk sent another PUBLISH telling that the call if terminated : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97ab8c7 52 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19863.16.0 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 230 <?xml version="1.0" encoding="UTF-8"?> terminated.. And Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97 ab8c752;received=192.168.100.37 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-48b4 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19873.18.1 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 Here, the SIP ETag is a.1518775074.19873.18.1. The problem is if I make a new call before the expiration of the previous SUBSCRIBE, Asterisk reuse this SIP ETag according to the RFC : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj9d13bb82-31d9-48db-9672-bd4b6b4f22 f0 From: sip:201@mydomain.com;tag=33e6b028-0444-4b3a-8bc2-4a987a291528 To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10699 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19873.18.1 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early. Kamailio refuse it with this error : "Trying to update an already terminated state. Skipping update." because the call is considered as terminated. The RFC is stating : When updating previously published event state, PUBLISH requests MUST contain a single SIP-If-Match header field identifying the specific event state that the request is refreshing, modifying or removing. This header field MUST contain a single entity-tag that was returned by the ESC in the SIP-ETag header field of the response to a previous publication. Why Kamailio is acting like that? Best regards, Cyrille

2 2

Problem with "transport=tcp" and rtpproxy
by Pau Giralt 16 Feb '18

16 Feb '18

Hi! I'm working with kamailio and rtpproxy. When I configure the clients to send sip messages through TCP adding the ";transport=tcp" in the sip message the rtpproxy is not working anymore. The code is never passing the point pasted below, inside route[NATMANAGE] in kamailio.cfg if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; Therefore, rtpproxy_manage("co"); is never reached. Could it be a parsing problem? ( check_route_param("nat=yes") ) The config file is exactly the same as the one provided in the kamailio repository (the default one) Thanks!

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Raspberry Pi deb packages -> built
by Mititelu Stefan 15 Feb '18

15 Feb '18

Hi people! After a bit of research and lib installations, I've managed to build the full kamailio deb packages for a raspberry pi 3 (version B) running raspbian stretch (lite) that can be found at [1] (with a really few module exceptions that might be used in really specific scenarios -> I think) Just for fun, I'm planning to do some CPS stats tomorrow (if I got the time), with a basic config and also share here, but, if someone else tried it before and is able to share some stats/benchmarks, would be greatly appreciated (and interesting to compare, given my "decent" rpi overclock), because I'm also curious about VoIP stuff on raspberry pi! :D [1] https://github.com/smititelu/rpi/tree/master/kamailio/5.1.1 Thank you, Stefan

2 3

KEMI
by Aleksandar Sosic 15 Feb '18

15 Feb '18

Hi everyone, we're valuating on rewriting our kamailio routes in python. We're doing a lot of string manipulation and GraphQL API queries so the kamailio scripting language is a little bit limited for this use case. Can someone tell us if the KEMI framework is stable enough for production? Kind regards, -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

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Re: [SR-Users] RTPEngine Transcoding
by Social Boh 15 Feb '18

15 Feb '18

Hello, I'm trying to use this configuration: Softphone A audio codec g722 and Opus Softphone B audio codec PCMA and PCMU Kamailio with rtpengine_manage with codec-transcode-PCMU codec-transcode-PCMA in the request Softphone A call Softphone B The RTPEngine DEBUG show many Decoder Error: https://pastebin.com/k1Zhw8HS Thank you Regards --- I'm SoCIaL, MayBe El 13/02/2018 a las 20:05, Richard Fuchs escribió: > On 2018-02-13 07:22 PM, Social Boh wrote: >> Thank you. >> >> which is the best way to implement ir >> >> codec-strip-all and the add codecs? >> >> Is there a document about how work the transcoding? > It's really up to you and depends on what you want to achieve. Doing a > strip-all and then adding codecs you want to offer is definitely one > way. Another is to just add codecs that you know or think the peer may > want to see. > > Cheers >

2 2

Invalid payload location
by Soltanici Ilie 15 Feb '18

15 Feb '18

Hi All, Can someone explain why i'm receiving such errors in syslog: [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:228]: set_sdp_payload_fmtp(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location This is how the SIP Packet is looking: 2018/02/15 10:51:40.916040 46.166.139.7:30189 -> X.X.X.X:5060 INVITE sip:01031203697179@X.X.X.X:5060 SIP/2.0 Via: SIP/2.0/UDP 46.166.139.7:30189;branch=z9hG4bK-5342365299746297160962993236828333618949;rport=30189 From: <sip:88893855391@46.166.139.7:30189>;tag=0vkn019bur To: <sip:01031203697179@X.X.X.X:5060> Contact: <sip:88893855391@46.166.139.7:30189> Call-ID: 0889633580294714035892119863939070195463 CSeq: 101 INVITE User-Agent: Hello Annonymous Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 356 v=0 o=SIP-Session 950029666 1307322426 IN IP4 46.166.139.7 s=SIP c=IN IP4 46.166.139.7 t=0 0 m=audio 5075 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=fmtp:9 bitrate=64000 a=rtpmap:104 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Thank You.

1 0

python_exec return 0 causes script hangs; return -1 is treated like 0
by Anthony Alba 15 Feb '18

15 Feb '18

5.1.1: if python_exec method returns 0, then the script hangs If the python_exec method return -1, then expressions treat it as if it returned zero. Somewhere there seems to be an offset by 1 issue. if (!python_exec("method")) { # if you return -1, the script does not hang but treated like zero } ################## Example 1: class kamailio(object): def hello(self, msg): KSR.info("hello, world\n") return 0 Calling: python_exec("hello") will cause script hang ################### Example 2: class kamailio(object): def hello(self, msg): KSR.info("hello, world\n") return -1 Calling python_exec("hello") if (!python_exec("hello")) { ## method returns -1 ## why did we get here? xlog("L_INFO","method return zero!\n"); } shows INFO: <script>: method return zero!

2 1

[IGNORE] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Apologies for the noise; it is working now. Not sure how I got the extra " 0" appended to the method arguments for python_exec(). AAlba

1 0

[BUG?] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Using 5.1, the python_exc args string is getting an extra [_0' appended When I call python_exec("method", "string"); In python: def method(self, msg, args): KSR.info("DBG: {} {} {}".format(msg, args)) This will show 'string 0'; instead of string. AAlba

1 0

RTPEngine Transcoding
by Social Boh 14 Feb '18

14 Feb '18

Hi List, I just read on the RTPEngine page the Proxy RTP now support transcoding: https://github.com/sipwise/rtpengine#transcoding is there a plan to support it? Regards -- --- I'm SoCIaL, MayBe

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sr-users February 2018