# \#sip **URL:** https://janus.discourse.group/tag/sip/1.md [Latest](https://janus.discourse.group/latest.md) · [Categories](https://janus.discourse.group/categories.md) · [Tags](https://janus.discourse.group/tags.md) --- ## [In response to a 'secure' offerless SIP Invite, the Janus WebRTC server is missing SDES in the offer SIP 200OK with SDP](https://janus.discourse.group/t/in-response-to-a-secure-offerless-sip-invite-the-janus-webrtc-server-is-missing-sdes-in-the-offer-sip-200ok-with-sdp/1757)
**Author:** [@Ritesh](https://janus.discourse.group/u/Ritesh)\ **Replies:** 3\ **Last updated:** [January 23, 2026, 5:47pm UTC](https://janus.discourse.group/t/in-response-to-a-secure-offerless-sip-invite-the-janus-webrtc-server-is-missing-sdes-in-the-offer-sip-200ok-with-sdp/1757 "2026-01-23T17:47:15Z")
Hi, I am using the Janus WebRTC Server version 1.2.4 for secure SIP calling. Issue short description: When the Janus WebRTC server receives the offerless SIP Invite from the secure SIP SBC, then the server is expected … --- ## [Janus Doesn't Detect Missed Calls, Using 3cx as the VoIP server](https://janus.discourse.group/t/janus-doesnt-detect-missed-calls-using-3cx-as-the-voip-server/1738)
**Author:** [@Izaack](https://janus.discourse.group/u/Izaack)\ **Replies:** 1\ **Last updated:** [November 1, 2025, 1:11pm UTC](https://janus.discourse.group/t/janus-doesnt-detect-missed-calls-using-3cx-as-the-voip-server/1738 "2025-11-01T13:11:24Z")
I’m using Janus as the gateway to connect to 3cx pbx which is basically my VoIP. If a call rings and it’s not answered or missed, 3cx IVR automatically plays but this to Janus is transalated to an answered event. How do … --- ## [A conference between multiple webrtc and sip participants?](https://janus.discourse.group/t/a-conference-between-multiple-webrtc-and-sip-participants/1723)
**Author:** [@bmwebrtc](https://janus.discourse.group/u/bmwebrtc)\ **Replies:** 5\ **Last updated:** [October 1, 2025, 8:28am UTC](https://janus.discourse.group/t/a-conference-between-multiple-webrtc-and-sip-participants/1723 "2025-10-01T08:28:40Z")
Is it possible to set up a video conference between WebRTC participants (via Janus) and SIP clients (via FreeSWITCH) without running a conference on FreeSWITCH? Specifically: Can SIP clients be represented as if they… --- ## [learn janus server](https://janus.discourse.group/t/learn-janus-server/71)
**Author:** [@baqer](https://janus.discourse.group/u/baqer)\ **Replies:** 2\ **Last updated:** [September 24, 2025, 6:46am UTC](https://janus.discourse.group/t/learn-janus-server/71 "2025-09-24T06:46:32Z")
Hello, I own a chat site and I want to get janus server service. I work on node js & npm & socket io --- ## [Audio Disappeared after 40 seconds](https://janus.discourse.group/t/audio-disappeared-after-40-seconds/1720)
**Author:** [@jalonzo](https://janus.discourse.group/u/jalonzo)\ **Replies:** 1\ **Last updated:** [September 19, 2025, 8:17am UTC](https://janus.discourse.group/t/audio-disappeared-after-40-seconds/1720 "2025-09-19T08:17:31Z")
When using WIFI it work perfectly but if using mobile data , unable to hear the audio after 40 seconds . Im using asterisk 22.1.1 and janus 1.3.1 --- ## [How can I transfer incoming calls from a SIP master session to a helper session?](https://janus.discourse.group/t/how-can-i-transfer-incoming-calls-from-a-sip-master-session-to-a-helper-session/1703)
**Author:** [@muyan](https://janus.discourse.group/u/muyan)\ **Replies:** 2\ **Last updated:** [August 27, 2025, 6:40am UTC](https://janus.discourse.group/t/how-can-i-transfer-incoming-calls-from-a-sip-master-session-to-a-helper-session/1703 "2025-08-27T06:40:52Z")
Is it possible to use the transfer request? However, this request requires a SIP address as a parameter. I’m not sure what will happen if I use the master session’s address when there’s a helper session. --- ## [Janus SIP plugin still using private IP in Contact header (AWS EC2 fixed public IP)](https://janus.discourse.group/t/janus-sip-plugin-still-using-private-ip-in-contact-header-aws-ec2-fixed-public-ip/1670)
**Author:** [@rbalbuquerque](https://janus.discourse.group/u/rbalbuquerque)\ **Replies:** 6\ **Last updated:** [July 11, 2025, 1:59pm UTC](https://janus.discourse.group/t/janus-sip-plugin-still-using-private-ip-in-contact-header-aws-ec2-fixed-public-ip/1670 "2025-07-11T13:59:45Z")
Hi everyone, We’re facing an issue where Janus SIP plugin keeps registering using the private IP (172.31.0.38) in the Contact: header, even though we’ve set up the correct public IP configuration. Environment Janus V… --- ## [How to specify the interval time for Janus to send register to SipServer?](https://janus.discourse.group/t/how-to-specify-the-interval-time-for-janus-to-send-register-to-sipserver/1652)
**Author:** [@muyan](https://janus.discourse.group/u/muyan)\ **Replies:** 6\ **Last updated:** [June 26, 2025, 6:55am UTC](https://janus.discourse.group/t/how-to-specify-the-interval-time-for-janus-to-send-register-to-sipserver/1652 "2025-06-26T06:55:01Z")
I saw from the packet capture that SipServer responds to register with expire=360, and then Janus will randomly send a register message within 360 seconds. Now I want to control the interval time for sending this message… --- ## [Outbound keep alive timeout](https://janus.discourse.group/t/outbound-keep-alive-timeout/1651)
**Author:** [@lakshmideepak](https://janus.discourse.group/u/lakshmideepak)\ **Replies:** 2\ **Last updated:** [June 25, 2025, 11:10am UTC](https://janus.discourse.group/t/outbound-keep-alive-timeout/1651 "2025-06-25T11:10:29Z")
i deployed janus in aws and im running it as service i can able to register with my sip credientials and i can make call it is working but after 30 sec im seeing a log … outbound(0x7427b40029b0): keepalive timeout… even … --- ## [Deploy janus-gateway server with SIP plugin inside Docker in AWS EC2](https://janus.discourse.group/t/deploy-janus-gateway-server-with-sip-plugin-inside-docker-in-aws-ec2/1517)
**Author:** [@ricardo\_kaka07](https://janus.discourse.group/u/ricardo_kaka07)\ **Replies:** 4\ **Last updated:** [April 28, 2025, 2:21pm UTC](https://janus.discourse.group/t/deploy-janus-gateway-server-with-sip-plugin-inside-docker-in-aws-ec2/1517 "2025-04-28T14:21:07Z")
I am trying to deploy janus-gateway-server(build from the latest version of main) with a proxy-service as a sidecar. The main idea is to hide janus-gateway-server from exposing all its rest endpoints to the outside world… --- ## [SIP Wrong state](https://janus.discourse.group/t/sip-wrong-state/1401)
**Author:** [@Kinsley](https://janus.discourse.group/u/Kinsley)\ **Replies:** 6\ **Last updated:** [March 14, 2025, 7:57am UTC](https://janus.discourse.group/t/sip-wrong-state/1401 "2025-03-14T07:57:58Z")
Hi ,I need help to understand this case where Janus responses with this error { "janus": "ack", "session\_id": 8286593828297388, "transaction": "m477s7q3RuySsRBoS3Mc3Wu0sl2RGyBpMAGTHH1y", "plugin": "janu… --- ## [SIP audio not hearing](https://janus.discourse.group/t/sip-audio-not-hearing/1525)
**Author:** [@venkathir.2005](https://janus.discourse.group/u/venkathir.2005)\ **Replies:** 1\ **Last updated:** [March 14, 2025, 4:07am UTC](https://janus.discourse.group/t/sip-audio-not-hearing/1525 "2025-03-14T04:07:04Z")
Hi, After call accepted from SIP voice is not audible. \[7321113943586715\] The DTLS handshake for the component 1 in stream 1 has been completed \[7321113943586715\] The DTLS handshake has been completed \[7321113943586… --- ## [Issue with SIP Calls on 3CX Queue number and Janus SIP Test](https://janus.discourse.group/t/issue-with-sip-calls-on-3cx-queue-number-and-janus-sip-test/643)
**Author:** [@rashid](https://janus.discourse.group/u/rashid)\ **Replies:** 4\ **Last updated:** [March 12, 2025, 10:45am UTC](https://janus.discourse.group/t/issue-with-sip-calls-on-3cx-queue-number-and-janus-sip-test/643 "2025-03-12T10:45:22Z")
Hi, I wanted to bring to your attention an issue I’ve been encountering with SIP calls in our setup. We are using a 3CX PBX and Janus for SIP communication. Recently, I set up a queue in 3CX and registered a SIP extens… --- ## [SIP Client port in INVITE message](https://janus.discourse.group/t/sip-client-port-in-invite-message/1217)
**Author:** [@Vitaliy](https://janus.discourse.group/u/Vitaliy)\ **Replies:** 2\ **Last updated:** [March 8, 2025, 9:18am UTC](https://janus.discourse.group/t/sip-client-port-in-invite-message/1217 "2025-03-08T09:18:05Z")
Good day, everyone! Then make call to SIP, i see INVITE message. For example: SummarySession Initiation Protocol (SIP as raw text) INVITE sip:45077@10.15.81.30 SIP/2.0\\r\\n Via: SIP/2.0/TCP 10.1.96.65:39995;branch=z9… --- ## [Enable DTLS-SRTP in janus-gateway server with sip plugin](https://janus.discourse.group/t/enable-dtls-srtp-in-janus-gateway-server-with-sip-plugin/1507)
**Author:** [@ricardo\_kaka07](https://janus.discourse.group/u/ricardo_kaka07)\ **Replies:** 2\ **Last updated:** [March 3, 2025, 10:25am UTC](https://janus.discourse.group/t/enable-dtls-srtp-in-janus-gateway-server-with-sip-plugin/1507 "2025-03-03T10:25:49Z")
I want to enable dtls-srtp in janus-server with sip plugin. I had set the self-assigned to true here. # Certificate and key to use for DTLS (and passphrase if needed). If missing, # Janus will autogenerate a self-sign… --- ## [DataChannels on SIP Plugin](https://janus.discourse.group/t/datachannels-on-sip-plugin/1435)
**Author:** [@edwinkr2000](https://janus.discourse.group/u/edwinkr2000)\ **Replies:** 6\ **Last updated:** [March 3, 2025, 9:55am UTC](https://janus.discourse.group/t/datachannels-on-sip-plugin/1435 "2025-03-03T09:55:06Z")
OpenAI recently added the ability to connect to its Realtime API via Webrtc, enabling realtime speech to speech AI apps. Other AI platforms will likely follow suit. The webrtc implementation on the Realtime API uses dat… --- ## [Missing remote audio on recording file](https://janus.discourse.group/t/missing-remote-audio-on-recording-file/1489)
**Author:** [@RoelfK](https://janus.discourse.group/u/RoelfK)\ **Replies:** 2\ **Last updated:** [February 19, 2025, 10:57am UTC](https://janus.discourse.group/t/missing-remote-audio-on-recording-file/1489 "2025-02-19T10:57:38Z")
Hi. Im wondering if someone can assist me with some strange behavior. Ive implemented a client app using SIP and the calls work perfectly with 2 way audio. Additionally i need the calls recorded so ive used the recorder … --- ## [Allow janus webrtc client to join an ongoing conference without registering first. Also need to authenticate using JWT](https://janus.discourse.group/t/allow-janus-webrtc-client-to-join-an-ongoing-conference-without-registering-first-also-need-to-authenticate-using-jwt/1471)
**Author:** [@ricardo\_kaka07](https://janus.discourse.group/u/ricardo_kaka07)\ **Replies:** 11\ **Last updated:** [February 17, 2025, 10:08am UTC](https://janus.discourse.group/t/allow-janus-webrtc-client-to-join-an-ongoing-conference-without-registering-first-also-need-to-authenticate-using-jwt/1471 "2025-02-17T10:08:04Z")
Hi, Our company is going to use Janus, so I am given this task to setup janus-sip-gateway plugin which will act as a gateway between web browser agents and clients on sip platform. I need to do two things- Allow clients… --- ## [Janus Sip Plugin Early Media Signalling](https://janus.discourse.group/t/janus-sip-plugin-early-media-signalling/31)
**Author:** [@yldrmali](https://janus.discourse.group/u/yldrmali)\ **Replies:** 12\ **Last updated:** [November 13, 2024, 12:04pm UTC](https://janus.discourse.group/t/janus-sip-plugin-early-media-signalling/31 "2024-11-13T12:04:35Z")
Hi everyone, I’m using Janus Sip plugin and it’s working flawlessly so far. I’m developing an application and I would like to see who’s in front of the door before a call is answered. I thought this could be achived vi… --- ## [When will HMAC be enabled?](https://janus.discourse.group/t/when-will-hmac-be-enabled/1364)
**Author:** [@iyad](https://janus.discourse.group/u/iyad)\ **Replies:** 1\ **Last updated:** [November 13, 2024, 12:00pm UTC](https://janus.discourse.group/t/when-will-hmac-be-enabled/1364 "2024-11-13T12:00:06Z")
When will HMAC-Signed token authentication be extended to include sip plugin? --- ## [Server-Side Audio Transmission to Janus Using Node.js SIP Plugin Fails: Assistance Needed with .wav/.opus File Streaming](https://janus.discourse.group/t/server-side-audio-transmission-to-janus-using-node-js-sip-plugin-fails-assistance-needed-with-wav-opus-file-streaming/1339)
**Author:** [@Kinsley](https://janus.discourse.group/u/Kinsley)\ **Replies:** 1\ **Last updated:** [October 25, 2024, 8:46am UTC](https://janus.discourse.group/t/server-side-audio-transmission-to-janus-using-node-js-sip-plugin-fails-assistance-needed-with-wav-opus-file-streaming/1339 "2024-10-25T08:46:10Z")
Hi, I’m using the SIP plugin to handle registration and other boilerplate tools. However, I’m struggling to send audio from a .wav or .opus file through Janus so it reaches the callee once the callee picks up or answers … --- ## [Pre accept state in SIP plugin?](https://janus.discourse.group/t/pre-accept-state-in-sip-plugin/1249)
**Author:** [@adnanel](https://janus.discourse.group/u/adnanel)\ **Replies:** 3\ **Last updated:** [September 26, 2024, 1:54pm UTC](https://janus.discourse.group/t/pre-accept-state-in-sip-plugin/1249 "2024-09-26T13:54:53Z")
We are trying to connect our janus instance to a 3rd party media server which is using ice-lite. With ice lite, the media server is only required to respond to STUN requests but not to make them, which is exactly what t… --- ## [ICE Error on SipTest](https://janus.discourse.group/t/ice-error-on-siptest/1196)
**Author:** [@Vitaliy](https://janus.discourse.group/u/Vitaliy)\ **Replies:** 2\ **Last updated:** [August 1, 2024, 3:52am UTC](https://janus.discourse.group/t/ice-error-on-siptest/1196 "2024-08-01T03:52:30Z")
Hi, everyone! I have a strange problem with Plugin Demo Sip Gateway. Janus and Coturn are installed on the server. Echotest - works great and fast. But Siptest can’t establish a connection and disconnects with error … --- ## [We are facing weird sip call dropouts for unknown reasons](https://janus.discourse.group/t/we-are-facing-weird-sip-call-dropouts-for-unknown-reasons/1139)
**Author:** [@Shivansh](https://janus.discourse.group/u/Shivansh)\ **Replies:** 4\ **Last updated:** [July 1, 2024, 8:44am UTC](https://janus.discourse.group/t/we-are-facing-weird-sip-call-dropouts-for-unknown-reasons/1139 "2024-07-01T08:44:24Z")
Hi Janus team, Thanks for this wonderful project, we use sip plugin to make calls through our system and we make sure each user use dedicated sip credentials to make a calls while call works great for the most part but t… --- ## [SIP Sound Problem](https://janus.discourse.group/t/sip-sound-problem/1073)
**Author:** [@palimo](https://janus.discourse.group/u/palimo)\ **Replies:** 0\ **Last updated:** [May 28, 2024, 4:08am UTC](https://janus.discourse.group/t/sip-sound-problem/1073 "2024-05-28T04:08:25Z")
i have problem, I use the sip plugin for calls but I can’t get sound received for both sides, for call successfully but not for audio, can anyone help me? --- ## [Problem while trying to change udp port for sip plugin](https://janus.discourse.group/t/problem-while-trying-to-change-udp-port-for-sip-plugin/1006)
**Author:** [@Zqsd](https://janus.discourse.group/u/Zqsd)\ **Replies:** 0\ **Last updated:** [April 17, 2024, 8:46am UTC](https://janus.discourse.group/t/problem-while-trying-to-change-udp-port-for-sip-plugin/1006 "2024-04-17T08:46:26Z")
Hello, I’m having trouble modifying the RDP port range in Janus. I’m trying to set the range in janus.jcfg from 10,000 to 20,000 and in janus.plugin.sip from 20,000 to 40,000. However, the application keeps allowing ra… --- ## [Problem with audio in sip call](https://janus.discourse.group/t/problem-with-audio-in-sip-call/1005)
**Author:** [@Zqsd](https://janus.discourse.group/u/Zqsd)\ **Replies:** 0\ **Last updated:** [April 17, 2024, 8:10am UTC](https://janus.discourse.group/t/problem-with-audio-in-sip-call/1005 "2024-04-17T08:10:26Z")
I experience two-way audio every time on Firefox, but in Chrome and edge, I only get microphone access the first time I open the browser and grant permission. Here are the first time on chrome : Between regular Chro… --- ## [Problem While Changing Media IP Address](https://janus.discourse.group/t/problem-while-changing-media-ip-address/993)
**Author:** [@Zqsd](https://janus.discourse.group/u/Zqsd)\ **Replies:** 0\ **Last updated:** [April 12, 2024, 10:02am UTC](https://janus.discourse.group/t/problem-while-changing-media-ip-address/993 "2024-04-12T10:02:20Z")
Hello, I’m working with a Janus server (local server without a public IP address) to facilitate SIP calls behind a NAT. I’m also using a separate web server as the frontend with the SIP gateway demo. I believe I may ha… --- ## [Janus SIP, WebRTC transfer](https://janus.discourse.group/t/janus-sip-webrtc-transfer/968)
**Author:** [@ydg](https://janus.discourse.group/u/ydg)\ **Replies:** 8\ **Last updated:** [April 5, 2024, 3:50pm UTC](https://janus.discourse.group/t/janus-sip-webrtc-transfer/968 "2024-04-05T15:50:08Z")
Hi, Currently I use Janus with its (amazing) SIP plugin to bridge my PBX to WebRTC devices running my app. This works great, even over LTE connections. We use a coturn TURN server to relay RTP traffic. Suppose that I w… --- ## [Limiting incoming SIP calls in Janus SIP plugin to registered server only](https://janus.discourse.group/t/limiting-incoming-sip-calls-in-janus-sip-plugin-to-registered-server-only/853)
**Author:** [@ych](https://janus.discourse.group/u/ych)\ **Replies:** 1\ **Last updated:** [February 8, 2024, 10:22am UTC](https://janus.discourse.group/t/limiting-incoming-sip-calls-in-janus-sip-plugin-to-registered-server-only/853 "2024-02-08T10:22:28Z")
Hello everyone, I’m currently facing a challenge with a standard Janus setup, where we use the Janus WebSockets Interface along with the Janus SIP plugin. This setup enables app users to register on a SIP server, facili… [Next page](https://janus.discourse.group/tag/sip/1.md?match_all_tags=true&page=1&tags%5B%5D=sip)