Oops, error creating inbound SRTP session for component 1 in stream 1 #2024
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No, your issue is different: It's a libsrtp error in |
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That solved the problem! great! I need to learn more about the different components and their mission, any recommendations to read? I downgraded libsrtp to version 2.2.0. Thanks a lot |
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But that's not good news... it means that libsrtp 2.3.0 breaks something, apparently. I'll add it to the things to check. |
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Oh! sorry :-) Should I open a new issue or reopen this one? I might be able to do more tests these days if needed. |
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No need, I'll try and investigate myself, and reopen in case. If it turns out we need some more testing from who can replicate the issue, I'll make sure to ping you, thanks 🙂 |
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FWIW I've been successfully using libsrtp 2.3.0 for the last months. |
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In that case, @gdiazlo you may want to try with 2.3.0 again, but without using the LetsEncrypt certificates. For DTLS you don't need valid certificates at all, it's always self-signed in browsers as well, and if you don't provide any Janus will create one for you at startup. |
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Where should I put the certificates if I do not want to use a proxy? I just put them in all certificate blocks in the config files. |
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The transport configuration file: the certificates setting in janus.jcfg is, as explained in the commented section, only for DTLS. |
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I tested with the certificate block and lets encrypt certificates only in janus.transport.http.jcfg and janus.transport.websockets.jcfg. No certificates in janus.jcfg. Libsrtcp 2.3.0 and janus 0.9.1. Still has the behaviour described in the issue. [ERR] [dtls.c:janus_dtls_srtp_incoming_msg:870] [2099492507476063] Oops, error creating inbound SRTP session for component 1 in stream 1?? @atoppi have you tested it with firefox? which version? It fails to me with firefox 74 :( Thanks! |
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I have the same issue on two instances of Janus, and can help test as needed. |
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in case it is relevant: I run Debian testing on armhf hardware. |
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compiling libsrtp2 with configure option @gdiazlo is your libsrtp2 linked with any of libssl or libnss, or without either of those? Libsrtp2 can optionally be linked with either libssl (a.k.a. openssl) or libnss, and it seems from the symbols of compiled code that without those a few GCM functions are missing. I notice if perhaps the build-time checks in janus for HAVE_SRTP_AESGCM should instead be runtime probing for libsrtp capabilities? |
I removed those checks. |
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Really? I mean lines like https://github.com/meetecho/janus-gateway/blob/5f75deb/dtls.c#L417 |
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HAVE_SRTP_AESGCM is always set to true right now. |
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ah. If janus cannot query at runtime which crypto ciphers are available, then I guess it would be helpful to at least document that janus needs libsrtp linked against libnss or libssl - because such linking is optional. |
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Had to get rid of that check because of #2019 which forced me to use pkg-config in #2033, and I can't check for methods using pkg-config. I asked for feedback on whether the check was needed, nobody answered, so I merged. We can't check at runtime, I think, because the method signature is always there, but the method is not always implemented. |
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I don't mean to say that any specific change to janus code is wrong - I am not qualified to judge that. I suggest to reopen this issue until janus can reliably probe for GCM support at runtime, and mention in README the requirement/recommendation of libsrtp with GCM capability. |
As I already said, you can't. Janus won't start because of undefined references, since the method is in the header, but is not implemented in some cases (I guess when libsrtp is not built against libssl or libnss). I had those checks in place exactly because I wanted to verify if they were in the library, but since I had to switch to pkg-config (and to my knowledge pkg-config doesn't have ways to check if a method is in the library, only if the library is there) I had to remove those checks. I asked for feedback and as usual nobody answered when I needed it, so here we are. I can add a note to the README, but won't investigate further, and will definitely not reopen this issue which is unrelated (again, if GCM isn't there it just won't start). If you guys have solutions, happy to review a pull request. |
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@jonassmedegaard it worked compiling in openssl or nss, or both. Thanks! |
Thanks. I think that would be helpful for your users.
I did not mean that you should solve this. What I meant was to keep this issue open until Cisco improves on that issue I referenced. But this is all just suggestions for how I think your users are best helped (not imposing extra work on you) - if you disagree then do whatever you like, this is your project :-) @gdiazlo thanks for confirming! |
Oh sorry, I missed the reference to the libsrtp issue. Makes sense now, thanks 👍 |
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Just added a |
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For reference, we have been observing Chrome 80 producing SPS units of 12 bytes or less. * Support for strings as unique IDs in AudioBridge, VideoRoom, TextRoom (meetecho#1880) * If glib is too old, generate uuid manually when needed (see meetecho#1880) * Fixed errors creating VideoRoom when strings are used (see meetecho#1880) * Remove duplicated codecs when answering SIP call (meetecho#1966) * Fixed a couple of JSON attributes in VideoRoom when strings are used (see meetecho#1880) * Make sure a publisher exists when asking for a VideoRoom subscriber renegotiation (fixes meetecho#1970) * Added errno info when socket operations fail in Streaming plugin * Fixed typos in TextRoom * Support for strings as unique mountpoint IDs in Streaming plugin (meetecho#1969) * fix meetecho#1967 (meetecho#1968) Fixed error callback not being invoked when an HTTP error happens trying to attach to a plugin * Added checks on nice_address_set_from_string (fixes meetecho#1973) * Fixed broken method signature in Streaming plugin when not using libcurl * Remove /root from the list of protected folders. 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(meetecho#2010) Co-authored-by: Michael Shiel <[email protected]> * http_transport: add NULL checks (meetecho#2012) Refs meetecho#2005 * Update media direction in SIP plugin if remote address is 0.0.0.0 ('hold' fix) (meetecho#2013) * Prepare RTCP Sender Reports by considering the last RTP timestamp sent. (meetecho#2007) * Track pending nack cleanup tasks and cancel them when freeing a stream. (meetecho#2014) * Fixed typo in janus.js error code (fixes meetecho#2018 * Reverted change on janus.js (see meetecho#2018) * Resolve mDNS candidates asynchronously with GResolver (see meetecho#1998) (meetecho#2004) * Reference count janus_request instances (meetecho#2020) Added better management of refcount on HTTP session when using it too, and refcount support to hanus_http_msg as well * Updates to mutex unlocking in textroom and videoroom plugins (meetecho#2026) * Updated Changelog (0.9.2) * Bumped to version 0.9.3 * Add Python aiortc-based functional testing. (meetecho#1971) * test_aiortc: cleanup (meetecho#2027) * Fixed missing refcount init for Admin API (fixes meetecho#2029) * Bumping back to 0.9.2 to re-tag * Updated changelog for 0.9.2 * Bumped to version 0.9.3 (again) * janus_http: return earlier if request is NULL (meetecho#2031) * Fixed janus-pp-rec build warnings when using ffmpeg >= 4.x * Fixed VideoRoom destroy not working when using strings * Fixed av_register_all deprecation check in post-processor * plugins: drop tautology (meetecho#2041) gateway is always set before initialized, so the latter is always true. * Don't set ICE credentials when parsing remote credentials (meetecho#2046) * Detect libsrtp(2) using pkg-config (fixes meetecho#2019) (meetecho#2033) * Added support for static Opus files to Streaming plugin (meetecho#2040) * Added support for generic metadata to Streaming mountpoints * Fixed printout of metadata in Streaming demo * Added notes on building libsrtp (see meetecho#2024) * Add configurable DSCP ToS for PeerConnections (meetecho#2055) * Always add remote candidates from the libnice loop (see meetecho#2045) (meetecho#2048) * Fixed Streaming destroy not working when using strings * Use refcount for Streaming plugin helper threads (meetecho#2039) * Added option to disable building AES-GCM support (see meetecho#2024 and meetecho#2054) * Fixed typo * Fixed outdated info in VideoRoom docs * Fixed syntax error in sample Streaming plugin configuration file * Support for additional constraints on screenshare media (meetecho#2043) * refactoring-clean up (const-var, semicolons, ===, etc.) (meetecho#2044) * Reference subscriber when handling related messages (see meetecho#2045) (meetecho#2061) * Added option to configure time needed to detect a missing simulcast substream (meetecho#2063) * Reverted isTrickleEnabled check in janus.js (fixes meetecho#2064) * Don't show warnings for rtx RTCP packets * Made libnice warning clearer, and upped suggested version (fixes meetecho#2069) * Add missing info to videoroom "list" response (meetecho#2068) * Use custom GSource to handle HTTP request timeouts (see meetecho#2062 and meetecho#2066) (meetecho#2075) * Define the libnice version string as extern in version.h (fixes gcc10 error) * Fixed AudioBridge create API not working properly when using string IDs * Fixed a few typos in AudioBridge errors * Fix copy-paste error in Streaming plugin docs * Fix libasan use after free in janus_videoroom_handler when events are enabled (meetecho#2091) * Added project to resources in the docs * Return mountpoint IP addresses, if a bind interface/IP was provided * Swap RR/SR Report Blocks if the first block contains rtx data. (meetecho#2089) * Add support for playback of audio files in AudioBridge (meetecho#2088) * Updated Changelog (0.9.3) * Bumped to version 0.9.4 * Fixed returned address when adding multicast Streaming mountpoints * More checks when hanging up VideoRoom subscriber (see meetecho#2087) (meetecho#2093) * Added new docker image to the resources in the docs * Updated AudioBridge documentation with new playback feature * Don't wait forever for candidates when half-trickling * Add some missing static declarations to HTTP and WS transports. Co-authored-by: Lorenzo Miniero <[email protected]> Co-authored-by: Agustin Polo <[email protected]> Co-authored-by: Yongje Lee <[email protected]> Co-authored-by: Alessandro Toppi <[email protected]> Co-authored-by: Sebastian Schmid <[email protected]> Co-authored-by: Imer Husejnovic <[email protected]> Co-authored-by: Oscar <[email protected]> Co-authored-by: Irek <[email protected]> Co-authored-by: Tristan Matthews <[email protected]> Co-authored-by: Jon Rafkind <[email protected]> Co-authored-by: kuekerino <[email protected]> Co-authored-by: Yurii Cherniavskyi <[email protected]> Co-authored-by: Meirza Arson <[email protected]> Co-authored-by: Groupboard <[email protected]> Co-authored-by: Cameron Lucas <[email protected]> Co-authored-by: hxl-dy <[email protected]> Co-authored-by: Alessandro Amirante <[email protected]> Co-authored-by: mp16 <[email protected]> Co-authored-by: Paul Zhang <[email protected]> Co-authored-by: Philipp Hancke <[email protected]> Co-authored-by: Sean DuBois <[email protected]> Co-authored-by: Ancor Gonzalez Sosa <[email protected]> Co-authored-by: Michael Shiel <[email protected]> Co-authored-by: Michael Shiel <[email protected]> Co-authored-by: agclark81 <[email protected]> Co-authored-by: Alex Pavlov <[email protected]> Co-authored-by: alexamirante <[email protected]> Co-authored-by: Federico Lorenzi <[email protected]>
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I confirm this issue with the stock version of libsrtp2 2.3.0 installed via |
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I've getting this error, Could you help me? [ERR] [dtls.c:janus_dtls_srtp_incoming_msg:923] [1982740767440321] Oops, error creating inbound SRTP session for component 1 in stream 1?? |
I also got this. |
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I realized that on my Ubuntu box, there is another libsrtp 2.3.0 installed. I uninstalled it then everything works again |
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i also got this issue. then i update the libsrtp with version 2.4.2, re-configure janus , everything is ok, now |
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Same problem here with libsrtp 2.3.0 on Ubuntu 20.04 LTS.. I installed the version 2.2.0 as mentioned in the README and that worked fine! Thanks! |
hello!
I have a setup with:
janus-gatewat 0.9.1
libnice 0.1.16
libsrtp 2.3.0
When running the videoroom demo in firefox 74 with janus default configuration modified to add let's encrypt certificates and more debug, the publishing of the video fails. The janus-gateway states:
The demo works fine with chromium based browsers using the same computer and networks. Also the demo in https://janus.conf.meetecho.com/videoroomtest.html works fine, so there should be something on my set up.
I have checked the ports and IPs. There is no NAT in the server, and no other failures in the log, besides some warnings about the sockets server not being enabled)
I can't figure out why it has an error creating the SRTP session. This may be related to #1688 ? That issue the most similar I've been able to find.
Any hints?
Thanks alot!
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