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| 15:33.09 | mjordan | file: what are your thoughts on ASTERISK-24807? |
| 15:33.29 | mjordan | IIRC, we typically have to create the field manually in the dialplan so that the Max-Forwards is decremented on each outbound leg |
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| 16:00.10 | file | mjordan, we have stuff which is supposed to set it |
| 16:00.17 | file | (it's configurable in pjsip.conf) |
| 16:02.55 | Kobaz | you guys are always more alive |
| 16:03.07 | Kobaz | i need to do splitting out individual sip/rtp steams from a pcap into individual files |
| 16:03.16 | Kobaz | maybe some wireshark type thing |
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| 16:04.44 | Kobaz | ooo oej, maybe you know... i know you do a lot of sip work |
| 16:05.01 | Kobaz | if you're alive |
| 16:09.12 | pabelanger | Kobaz, wireshark can do it |
| 16:09.19 | pabelanger | will filter per stream |
| 16:09.21 | pabelanger | and per direction |
| 16:10.51 | Kobaz | yeah |
| 16:10.55 | Kobaz | i know you can filter that stuff |
| 16:11.04 | Kobaz | i'm running through 'voip calls' processing now |
| 16:11.07 | Kobaz | it's like at 90% |
| 16:11.10 | Kobaz | took about an hour |
| 16:11.23 | Kobaz | i want individual pcap files to play with |
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| 16:46.43 | oej | kobaz what? |
| 16:47.49 | Kobaz | hey, splitting out sip+rtp from massive pcap files |
| 16:48.08 | Kobaz | i'm trying pcapsipdump, but it's not finding any actual calls, it's just dumping sip options packets |
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| 16:58.49 | oej | i always use pcapsipdump. |
| 16:58.54 | oej | Strange if it doesn't work. |
| 16:59.23 | oej | Does wireshark find any calls? |
| 17:22.34 | stefan27 | Two crash-dumps related to DTLS: Using chan_sip and local channel driver only. the sip peers who have dtls encryption all use webrtc over WSS: http://pastebin.com/uyQdZaRv and http://pastebin.com/YUhPvLBF One Segmentation fault one ABORT, both happend within the same day during medium load ~30-50 channels up |
| 17:23.34 | stefan27 | common for the two crashes are that they go through dtls_do_write() using openssl-1.0.1e-30.fc20.x86_64 rpm from fedora 20 repo. I'm trying to find out if the bug is in openssl or asterisk |
| 17:23.58 | file | there are issues on the issue tracker for that already, you may want to take a look |
| 17:24.30 | stefan27 | I did browse through all jiras mathing the search word DTLS quickly. some of them may be related |
| 17:26.33 | stefan27 | I noticed that if I upgraded to openssl-1.0.1e-41.fc20.x86_64 from 1.0.1e-30 i needed to patch asterisk with the patch resulting from ASTERISK-24711 |
| 17:30.21 | stefan27 | but if crashes are not openssls fault upgrading won't help, and may cause other problems in fact... Maybe ASTERISK-24651 is related? It says affects versions 11.15.0 but it is unresolved |
| 17:31.15 | stefan27 | I'm using 13.1.0 |
| 17:34.39 | stefan27 | is 13 and 11 very similar regarding res_rtp_asterisk and dtls? |
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| 17:42.36 | file | yes. |
| 18:02.44 | Kobaz | oej: wireshark finds many calls |
| 18:02.58 | Kobaz | oej: i turned up debugging and pcapsipdump is skipping 99% of the packets |
| 18:03.12 | Kobaz | looking at the code it looks like a bug in the port tracking |
| 18:10.34 | Kobaz | pcalsipdump is only dumping SIP OPTIONS |
| 18:10.36 | Kobaz | and that's it |
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| 18:49.22 | stefan27 | Yeah, it crashes after 100-1000 calls... I had core debug level 5 on this time. I'm not sure how to debug. This time the last asterisk-call was from __rtp_recvfrom() with rtp-instance 0x7fee314ce598. I grepped the debug-log for this memory address and it shows the following messages: http://pastebin.com/dPSyMiUX (core-dump at http://pastebin.com/f91XWfcU ) |
| 18:50.45 | stefan27 | do any of those messages seem odd? It first allocates port 8016 for RTP instance '0x7fee314ce598' then about 2 minutes later Allocated port 8058 for RTP instance '0x7fee314ce598' and crashes 2 seconds after that |
| 18:53.34 | stefan27 | Oh, my bad, there was a "Destroyed RTP instance '0x7fee314ce598'" in between that the grep missed |
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| 19:06.44 | Kobaz | do you guys know what filters i would use in wireshark to link a sip/sdp to rtp |
| 19:06.54 | Kobaz | sdp.owner.sessionid in the sdp? |
| 19:07.09 | Kobaz | should that match rtp.ssrc? |
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| 19:12.18 | pabelanger | Kobaz, if you have wireshark open, under telephony there is RTP->Show all streams |
| 19:12.23 | pabelanger | that's the easiest way I find them |
| 19:12.52 | pabelanger | then, I click analyze and then save payload |
| 19:14.20 | Kobaz | yeah |
| 19:14.22 | Kobaz | the problem is |
| 19:14.33 | stefan27 | my rtp streams dont show up there, but they are encrypted |
| 19:14.38 | Kobaz | that's hard to do when your file is like a gig and a half |
| 19:14.47 | Kobaz | stuff takes forever |
| 19:14.57 | Kobaz | i'm trying to write a thing to extract out sip/rtp for individual calls |
| 19:15.08 | Kobaz | pcalsipdump would have been great, but it's broken |
| 19:17.25 | Kobaz | i had all the voip calls processed |
| 19:17.32 | Kobaz | took about an hour |
| 19:17.36 | Kobaz | and then i went to play one and then wireshark crashed |
| 19:19.39 | coreyfarrell | Kobaz: when I'm dealing with large pcap's I first find a call I'm interested in (usually by exporting SIP only to a separate file). Then I use editcap on the original pcap to filter by time range of the call. |
| 19:20.05 | Kobaz | the problem is, i need to go through 35 calls |
| 19:27.32 | Kobaz | doesn't help wireshark keeps crashing |
| 19:33.35 | coreyfarrell | yeah wireshark does that to me with large files :( |
| 19:33.47 | Kobaz | this isnt even a large file anymore |
| 19:33.52 | Kobaz | i dumped a specific call finally |
| 19:33.55 | Kobaz | it's 50 megs |
| 19:34.22 | coreyfarrell | oh I've never had that happen. sorry |
| 19:36.11 | Kobaz | happens all the time for me |
| 19:36.17 | Kobaz | wireshark just likes to crash |
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| 20:04.13 | stefan27 | upgrading to openssl 1.0.1e-41 from -30 actually may have helped with my webrtc/dtls crashes in asterisk... Too early to tell for sure but it survived 1300 calls now and before the upgrade it crashed after 300, 700, 600 calls respectively, same test-case |
| 20:04.28 | stefan27 | Might be chance though -- hate nondeterministic bugs |
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| 23:15.53 | *** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Asterisk 13.0.0 now released. Do the release dance! -=- Follow on Twitter at @AsteriskDev |