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Error on transfer when calling from WebRTC with TOPOH
by Arsenijs Kabihno 15 Mar '23

15 Mar '23

Hello, We have an issue which appears when TOPOH module is enabled, call is made from WebRTC, then transferred by calee: Mar 15 11:55:27 ap07stvp.xxxxxxx.xxx docker[2062]: 29(35) INFO: {2 2 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bK66807520, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address via-branch=extra rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Mar 15 11:55:27 ap07stvp.xxxxxxx.xxx docker[2062]: 29(35) NOTICE: {2 2 INVITE g2rgvnioadg8m7l6i7k6} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1678881327;method=INVITE;from_tag=p9qqi7hrua;to_tag=ea7d2850-3e76-4c11-821c-a0d1e3cbb011;call_id=g2rgvnioadg8m7l6i7k6;code=200;reason=OK;src_user=92010;src_domain=10.181.0.1;src_ip=10.5.0.145;dst_ouser=27221;dst_user=27221;dst_domain=10.181.0.1 Mar 15 11:55:38 ap07stvp.xxxxxxx.xxx docker[2062]: 31(37) INFO: {1 27689 BYE e034a187-0e4d-4c4e-ab4b-f2e5f1281330} <script>: RTPENGINE branch_id:0 ruri: sip:97093@10.128.150.67:46058;rinstance=d403efb922420649;transport=UDP, method:BYE, status:<null>, extra_id: <null>, rtpengine_delete Mar 15 11:55:38 ap07stvp.xxxxxxx.xxx docker[2062]: 28(34) NOTICE: {2 27689 BYE e034a187-0e4d-4c4e-ab4b-f2e5f1281330} acc [acc.c:287]: acc_log_request(): ACC: transaction answered: timestamp=1678881338;method=BYE;from_tag=0c29be45-6aff-4d4e-8503-650893a5a5ac;to_tag=b1a8a401;call_id=e034a187-0e4d-4c4e-ab4b-f2e5f1281330;code=200;reason=OK;src_user=22628;src_domain=10.5.2.192;src_ip=10.5.2.192;dst_ouser=97093;dst_user=97093;dst_domain=10.128.150.67 Mar 15 11:55:39 ap07stvp.xxxxxxx.net docker[2062]: 32(38) INFO: {1 27798 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: sip:jgqtuev6@2idi5cucan2v.invalid;transport=ws;ob, method:INVITE, status:<null>, extra_id: <null>, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 SIP-source-address direction=priv1 direction=pub1 rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 27(33) INFO: {1 4728 INVITE 133ccd4c-031b-4d10-b073-745f35433132} <script>: RTPENGINE branch_id:0 ruri: sip:0037126048570@193.138.189.220, method:INVITE, status:<null>, extra_id: z9hG4bKPjedcf2d96-01cd-43fd-bed0-4e601e3e542f0, rtpengine_offer: replace-origin replace-session-connection SIP-source-address direction=priv1 direction=pub1 via-branch=extra Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 50(56) INFO: {2 27798 INVITE g2rgvnioadg8m7l6i7k6} <script>: RTPENGINE branch_id:0 ruri: <null>, method:INVITE, status:200, extra_id: z9hG4bKPj1cd87bda-7862-4ca9-8475-ac6b342c45ef0, rtpengine_answer: replace-origin replace-session-connection SIP-source-address direction=pub1 direction=priv1 SIP-source-address via-branch=extra rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} <core> [core/resolve.c:1730]: sip_hostport2su(): could not resolve hostname: "2idi5cucan2v.invalid" Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} <core> [core/forward.c:516]: forward_request(): bad host name 2idi5cucan2v.invalid, dropping packet Mar 15 11:55:39 ap07stvp.xxxxxxx.xxx docker[2062]: 25(31) ERROR: {1 27798 ACK g2rgvnioadg8m7l6i7k6} sl [sl_funcs.c:418]: sl_reply_error(): stateless error reply used: Unresolvable destination (478/SL) We have the following setup: WebRTC client behind load-balanser -> kamailio -> asterisk. When call is transferred, asterisk makes re-invite. If TOPOH is disabled, it happens this way and issue is not present: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> WebRTC ACK example: 03/09/2023 09:35:48.126 +02:00: 10.5.2.192:5060 -> 10.181.0.1:5060 ACK sip:98dn3met@0j62ln34srjn.invalid;transport=ws;ob;alias=10.5.0.145~27562~6 SIP/2.0 Via: SIP/2.0/UDP 10.5.2.192:5060;rport;branch=z9hG4bKPj1d11dae2-10bf-466e-9221-5710f801d725 From: <sip:27221@10.181.0.1>;tag=b6b73135-efe1-4d86-bbc8-bf4313130eb5 To: <sip:92010@10.181.0.1>;tag=namgu127j9 Call-ID: t0hmmk50h5i4ok88asjj CSeq: 28632 ACK Route: <sip:10.181.0.1:5060;lr;r2=on;ftag=namgu127j9;did=aaa.7b92;rtp=bridge;rtp=ws> Route: <sip:10.181.0.1:4444;transport=ws;lr;r2=on;ftag=namgu127j9;did=aaa.7b92;rtp=bridge;rtp=ws> Max-Forwards: 70 User-Agent: Asterisk PBX 18.16.0 Content-Length: 0 If TOPOH is enabled, kamailio doesn't pass this ACK to WebRTC: asterisk INVITE -> kamailio -> WebRTC WebRTC 200 -> kamailio -> asterisk asterisk ACK -> kamailio -> -> -> ACK example: 03/09/2023 09:33:03.544 +02:00: 10.5.2.192:5060 -> 10.181.0.1:5060 ACK sip:10.0.0.1;line=Bn7Ipzh8eb8DxMSFyk.rWzitxzVFBoirxRG5xYeOxb7hpoXTlMGDBbuTCk3oBDAqlH** SIP/2.0 Via: SIP/2.0/UDP 10.5.2.192:5060;rport;branch=z9hG4bKPjf00fc541-f011-4e52-b63a-2c569478ca39 From: <sip:27221@10.181.0.1>;tag=7d402872-7355-4269-b912-e52ee28ed415 To: <sip:92010@10.181.0.1>;tag=t0q4g9am9g Call-ID: t0hmmetickvlvn5k7nco CSeq: 25565 ACK Route: <sip:10.181.0.1:5060;lr;r2=on;ftag=t0q4g9am9g;did=4c8.a182;rtp=bridge;rtp=ws> Route: <sip:10.0.0.1;line=Bn7IpzdIwzd8Vf8Iwzd9WkyFWkAFBELJBo.qBYyuCoVgBz1uxn8gx61uxn8geYXOeD3FV6dFeD7Oxj7YpnX5ekFFlDHJljd8VzATC60ulYi5ebC7poiFBk3oBI**> Max-Forwards: 70 User-Agent: Asterisk PBX 18.16.0 Content-Length: 0 We have an assumption that issue might be related to the way how TOPOH handles alias in contact. Did someone have encountered something like this?

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Advertise NAT IP address causing Multiple SIP INVITEs is that expected ?
by Adil Mafzool Zaka 15 Mar '23

15 Mar '23

Hi I am using advertise in the listen interface for our NATed setup within AWS. Using following listen statement in kamailio: - version: kamailio 5.5.1 listen=udp:10.0.1.4:5060 advertise 3X.1XX.1XX.2XX:5060 dispatcher.list 1 sip:10.0.4.23:5060 8 10 weight=11 #FreeSwitch 1 We are seeing that kamailio (10.0.1.4) is sending two SIP INVITEs via dispatcher to FreeSwitch at 10.0.4.23 which seems strange or is it expected behaviour? (As seen below) SIP INVITE Request: - WAN Address >> Kamailio AWS NATed address >> FreeSwitch 1 1X.2X.2X.2X >> 3X.1XX.1XX.2XX 10.0.1.4 >> 10.0.4.23 10.0.1.4 >> 10.0.4.23 Many thanks Regards Muhammad Zaka

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Use Kamailio and RTPEngine to play wav file as early media to announce a server maintenance
by jhayhoe＠liquidweb.com 15 Mar '23

15 Mar '23

I am looking to configure Kamailio and RTPEngine to play a wav file for any new calls during a scheduled asterisk server maintenance. I have set up the following within route{} to send the call to a early_media route based on a variable that I can set on the server using kamcmd. route { ... if ($sht(config=>maint)=="1") { if (is_method("INVITE") && !has_totag()) { xlog ("L_INFO","Ongoing maintenance. Call-ID:$ci | Method: $rm"); route(early_media); } } ... } route[early_media] { xlog("Maintenance: handling early_media\n"); rtpengine_manage("direction=external direction=external"); $var(body) = $(rb{re.subst,/(IP4.).*/\1##server IP redacted##/g}); set_reply_body($var(body), "application/sdp"); t_send_reply("183", "Session Progress"); sleep(2); unblock_dtmf("all"); play_media("file=/etc/kamailio/media/maintenance.wav"); sleep(21); rtpengine_delete(); t_reply("480","Temporarily Unavailable"); } This mostly works as I hear the recorded message in place of the ringing tone when a call is processed by kamailio and then a busy signal is heard as part of the 480 reply at the end. The sleep's are in place to match the duration of the recorded message as otherwise the 480 is sent immediately following the start of the play_media. Feel free to let me know if there is a better solution to wait the duration of the media file and then proceed with processing instead of using sleep(). Is there a way to instead send a CANCEL or a BYE instead of a 480 after the wav file is played? I tried using t_cancel_callid("$ci", "$cs", "0") as well as a dlg_bye("all") at the end of the early_media route and neither seemed to send any headers out.

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stuck with webrtc on kamailio
by Arun K R 14 Mar '23

14 Mar '23

hello, I have issue of no voice on kamailio configured for webrtc. I have configured rtpengine with public and private interface. My kamilio has two interface of pubic and private ip. Somebody has suggested me that there is no ice trickling happens. I am using jssip webrtc client. Kindly suggest me or help me with this. I have send another email for same issue to this group and didnt get any reply. Kindly help me

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Re: Your message to sr-users@lists.kamailio.org awaits moderator approval
by Arun K R 14 Mar '23

14 Mar '23

I need a help on issue with rtpengine configuration and I didn't get any repy for it. Kindly check about it and help me. If i won't get any help kindly let me know, so I can stop waiting for your reply On Mon, Mar 13, 2023 at 12:50 PM <sr-users-bounces(a)lists.kamailio.org> wrote: > Your mail to 'sr-users(a)lists.kamailio.org' with the subject > > webrtc audio issue > > Is being held until the list moderator can review it for approval. > > The message is being held because: > > The message is larger than the 150 KB maximum size > > Either the message will get posted to the list, or you will receive > notification of the moderator's decision. >

1 0

send messages from source ip in Linux anyIP list
by mohsen khashei 14 Mar '23

14 Mar '23

1 0

Dispatcher Incoming
by nutxase 14 Mar '23

14 Mar '23

Hi All I have setup kamailio using dispatcher to proxy registrations from the UAC to asterisk but when asterisk sends an incoming call it does not seem to keep the path header and therefore kamailio sends 404 is there anyway around this? Thanks Sent with [Proton Mail](https://proton.me/) secure email.

5 18

Use MySQL connector instead of mariadb connector
by suchendra adiga 14 Mar '23

14 Mar '23

Hi, I have been working on setting up the database with the Kamailio server. But facing a problem when my db user has configured with SSL required. But I learnt that there is no support to enable the SSL in mariadb connector. So I am thinking of using the MySQL connector instead. Is there anyway I could configure the kamailio to use MySQL instead of Mariadb? Thanks and regards, Suchendra

2 7

Advertise NAT IP address causing Multiple SIP INVITEs is that expected ?
by Muhammad Zaka 13 Mar '23

13 Mar '23

Hi We are using advertise in the listen interface for our NATed setup within AWS. Using following listen statement in kamailio: - version: kamailio 5.5.1 listen=udp:10.0.1.4:5060 advertise 3X.1XX.1XX.2XX:5060 dispatcher.list 1 sip:10.0.4.23:5060 8 10 weight=11 #FreeSwitch 1 We are seeing that kamailio (10.0.1.4) is sending two SIP INVITEs via dispatcher to FreeSwitch at 10.0.4.23 which seems strange or is it expected behaviour? (As seen below) SIP INVITE Request: - WAN Address --> Kamailio AWS NATed address --> FreeSwitch 1 1X.2X.2X.2X --> 3X.1XX.1XX.2XX 10.0.1.4 --> 10.0.4.23 10.0.1.4 --> 10.0.4.23 [cid:image001.png@01D955CE.9F0C2EE0] Many Thanks Regards Muhammad Zaka Muhammad Zaka UCaaS Solutions Engineer Tel: 03330147908 Email: Muhammad.Zaka(a)gamma.co.uk This is an email from Gamma Telecom Ltd, trading as “Gamma”. The contents of this email are confidential to the ordinary user of the email address to which it was addressed. This email is not intended to create any legal relationship. No one else may place any reliance upon it, or copy or forward all or any of it in any form (unless otherwise notified). If you receive this email in error, please accept our apologies, we would be obliged if you would telephone our postmaster on +44 (0) 808 178 9652 or email postmaster(a)gamma.co.uk Gamma Telecom Limited, a company incorporated in England and Wales, with limited liability, with registered number 04340834, and whose registered office is at The Scalpel, 18th Floor, 52 Lime Street, London EC3M 7AF and whose principal place of business is at Kings House, Kings Road West, Newbury, Berkshire, RG14 5BY.

1 0

(no subject)
by gui 10 Mar '23

10 Mar '23

Hello I have an issue with Rx in my VolTE call Flow. Please refer to picture and pcap in annex. I am using same rtp engine at origination and termination. I am sending rtpmanage for invite and 183 session progress at both origination and termination. My issue is that Rx is not sending for the proper media flow component to PCRF. At origination it is considering a media flow between A 50034 and RTPEngine 49186 (instead of considering RTP Engine Port 49152) At Termination it is considering a media flow between B 50054 and RTPEngine 49132 (instead of considering RTP EnginePort 49170) Any idea why?

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