Comments (5)
I still don't understand how SIP messages can get that large, when the SIP plugin doesn't even support multistream and so can only do two m-lines. Are you using event handlers (which is the only thing using those log buffers) or not (in which case we'd need to check why the errors appear in the first place)?
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Hello lminiero, I break point here found that the judgment "if(line[3] == '-')" has never been hit, and then "sofia_log[0] = '\0';" will never be executed to clear the buffer.
After executing several more sip operations, sofia_log will be full. If continue to execute sip operations, the error "Truncation occurred" will be reported.
I don’t understand sofia’s log structure. Should every log received be forwarded to the event handler without making this judgment?
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I copied the value of 'line' with a registered sip message. It is like this:
0x7fffb67eb270 "recv 361 bytes from udp/[192.168.19.59]:5060 at 14:54:04.204265:\n", '-' <repeats 72 times>, "\nSIP/2.0 200 OK\nVia: SIP/2.0/UDP 192.168.9.245:32868;rport=32868;branch=z9hG4bK3Bet6r1mgNp7p;received=192.168.9.245\nFrom: <sip:[email protected]>;tag=3345Fm7Fj4vQB\nTo: <sip:[email protected]>;tag=cd1747d9\nCall-ID: 6eec95a2-86e1-123d-768e-aa22c5774c0e\nCSeq: 82978630 REGISTER\nContact: \"3827\"<sip:[email protected]:32868;transport=udp>;expires=360\nContent-Length: 0\n\n\n"
the character -
is in bits 62 to 133. I guess the judgment of line[3] == '-'
is actually Want to judge these characters?
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I suspect that the log output may be related to the sofia-sip version. Currently, I am using version 1.13.17.
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Yes, I think so too: there's was an issue some time ago that was opened about that, #3031. The user that opened the issue contributed a tentative PR , but then disappeared so it never went anywhere. If you have ways to replicate the problem and maybe revisit the original PR (#3063) to see if it works for you, it would help having an up-to-date fix for that.
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Related Issues (20)
- [1.x] Sip session get stuck HOT 12
- No call back triggering in 'onlocalstream' HOT 2
- [1.x] Streaming plugin does not send "started" event to event handlers HOT 2
- Add unauthenticated no-payload ping endpoint for health checks HOT 3
- Build fails with latest libsrtp version HOT 9
- [0.x] [janus.js] black screen or streaming stops periodically when connected through WebRTC (Chromium only) HOT 4
- [1.x] Issue with outbound calls for SIP Gateway HOT 3
- [1.x] videoroom: remote publisher doesn't release RTP/RTCP ports HOT 10
- [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 14
- [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 7
- [1.x] videoroom: support of string_ids/string_ids_user HOT 2
- [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
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