Comments (6)
No, all my sip replies were 200 OKed, the known and unknown ips.
Mh, maybe it shouldn't be 200 for unknown IPs, but we'll see in the future. Thanks for the feedback!
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It looks like an issue looking up the To
in the hashtable. I see the OPTIONS has the To
set to
To: <sip:JANUS_IP:5080>.
This means that our
const char *to = sip->sip_to->a_url[0].url_user;
is probably NULL
or empty, which makes the g_hash_table_lookup(sip_trunk->sessions, to)
call crash. I'll try to improve that check, so that you can see if it fixes things for you.
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@do-not-set-2fa can you check the commit above?
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Hey Lorenzo,
It is indeed fixed, surviving already 10min of sip options hammering (6 messages per minute :)
Options for trunk type are a bit loose, no need for proper to i guess.
Thanks for the quick fix, we will do more poc/testing on this branch soon i guess.
One unrelated question, maybe more for a discord:
Now that there is a trunk, can there be a bypass media mode?
For example you have another sip sbc that can talk to user so no need for janus to ne in the media path (less jumps, less delay and potential breakpoints), just to do webrtc json to sip signaling conversion and keep original sdp in offer and answer intact?
Rtpengine does the kernel udp forwarding and can bring it back to user space when needed, so it is much faster and can handle more media streams that way. (Also would be wery cool to have kernalization in janus ofcourse, that would be even better, set up a stream in user space and move it to kernel space if there is no need for transcoding and such)
Thanks,
Mirko
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It is indeed fixed, surviving already 10min of sip options hammering (6 messages per minute :)
Is Janus sending a 200 in response to those OPTIONS, or a 404? If it's a 404, there still might be something we need to fix I guess.
Now that there is a trunk, can there be a bypass media mode?
No, Janus and the SIP plugin will alwaye be on the media path.
Also would be wery cool to have kernalization in janus ofcourse
I don't think this is viable, due to the general purpose nature of Janus. Different plugins will need to handle media in different ways, there's not just a core relaying media from A to B and viceversa. As such, it's not planned for now.
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No, all my sip replies were 200 OKed, the known and unknown ips.
Olk thanks for the info and fast replies as always :)
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Related Issues (20)
- [1.x] Does AudioBridge supports multiple participant join from RTP in a single room . Getting `Already in a room` Error for 2nd participant HOT 3
- [1.x] rtp_port_range not being respected HOT 2
- [1.x] Crash / segfault on 1.2.2 HOT 5
- [1.x] system under heavy load stops to handle calls wia ws api HOT 19
- [1.x] Crash / segfault on 1.2.3 HOT 10
- [1.x] inconsistency happen when more than 6 users join at same time HOT 2
- [1.x]inconsistency happen when more than 6 users join at same time with different browsers (web and mobile) HOT 10
- [1.x] videoroom: support of string_ids/string_ids_user HOT 2
- [1.x] Increased event handlers SIP logging buffer size one more time or make it configurable HOT 7
- [1.x] build failure on macos HOT 3
- [0.x] Possible memory leaks found with libasan latest HOT 12
- [1.x] SIP plugin "Couldn't connect audio RTP? -- 22 (Invalid argument)" HOT 5
- [1.x] Transport Websockets plugin binds to ANY network interface HOT 4
- [1.x] AudioBridge Talking Event is not firing HOT 8
- Failed to compile janus on an arm machine inside docker container with ubuntu 24.04 as image HOT 1
- [1.x] Im used Janus in docker contaner with docker-compose HOT 1
- [1.x] Commandline option "--version" no longer exists HOT 1
- [1.x] why is it recommended to use libnice v0.1.18? libnice 0.1.19 exists a number of bug fixes HOT 1
- [1.x] HOT 1
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