> On Feb. 24, 2014, 4:51 p.m., Joshua Colp wrote: > > /branches/11/res/res_rtp_asterisk.c, lines 554-560 > > <https://reviewboard.asterisk.org/r/3256/diff/1/?file=54416#file54416line554> > > > > Agreed, and pjnath does not provide a mechanism to do just that without > > destroying/re-creating as Matt says. > > > > What would also be useful is to further look at the SDPs involved - are > > the candidates really changing? Do we really need to restart the ICE > > negotiation? > > Jonathan Rose wrote: > No, the candidates offered aren't changing. I think the reason it tries > to add them anyway is because we never reached a point where the ICE session > was tracked as having started (because the check list compliation is failing). > > > First Invite (SIPML5 client to desk phone) > > <--- SIP read from WS:10.24.16.82:60366 ---> > INVITE sip:1201@10.24.18.246 SIP/2.0 > Via: SIP/2.0/WS > df7jal23ls0d.invalid;branch=z9hG4bKMKg6LeUkDugGr9B1CnyHICANfn8JwRVR;rport > From: "sipml_bot"<sip:sipml_bot@10.24.18.246>;tag=W42MEkWLdFTUiIaaw8iy > To: <sip:1201@10.24.18.246> > Contact: > "sipml_bot"<sip:sipml_bot@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr" > Call-ID: 97982d1a-15e5-cd55-9122-07b3cc20cb7d > CSeq: 20907 INVITE > Content-Type: application/sdp > Content-Length: 1840 > Max-Forwards: 70 > User-Agent: IM-client/OMA1.0 sipML5-v1.2014.01.27 > Organization: Doubango Telecom > > v=0 > o=- 3044420497219628500 2 IN IP4 127.0.0.1 > s=Doubango Telecom - chrome > t=0 0 > a=group:BUNDLE audio > a=msid-semantic: WMS nqy2BUfoo7yn1YeWLNBeED2rqMGSb1y8Ypv3 > m=audio 56984 RTP/SAVPF 111 103 104 0 8 106 105 13 126 > c=IN IP4 216.207.245.1 > a=rtcp:56984 IN IP4 216.207.245.1 > a=candidate:474352566 1 udp 2113937151 10.24.16.82 56984 typ host > generation 0 > a=candidate:474352566 2 udp 2113937151 10.24.16.82 56984 typ host > generation 0 > a=candidate:3038348387 1 udp 1845501695 216.207.245.1 56984 typ srflx > raddr 10.24.16.82 rport 56984 generation 0 > a=candidate:3038348387 2 udp 1845501695 216.207.245.1 56984 typ srflx > raddr 10.24.16.82 rport 56984 generation 0 > a=candidate:1388705606 1 tcp 1509957375 10.24.16.82 0 typ host generation > 0 > a=candidate:1388705606 2 tcp 1509957375 10.24.16.82 0 typ host generation > 0 > a=ice-ufrag:LeGzZojC7gQNvl7o > a=ice-pwd:VfqsY2VVKr3xg/mgvdPtcxhp > a=ice-options:google-ice > a=fingerprint:sha-256 > 3A:BA:F5:44:53:CD:99:6C:D1:32:9F:80:53:D4:B5:BA:AE:CF:98:54:71:7F:D6:CB:14:7F:D8:94:30:98:89:62 > a=setup:actpass > a=mid:audio > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=sendrecv > a=rtcp-mux > a=crypto:0 AES_CM_128_HMAC_SHA1_32 > inline:ZQQ2MjApfbPlEAmWe52FRkbKUVH4rXt5p5QnpObB > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:JljeDW3NiakHNYn+bu+vje3cD77nNPytUW79J2Vz > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=maxptime:60 > a=ssrc:3822531142 cname:YE/Lm1aHvDjHxwH0 > a=ssrc:3822531142 msid:nqy2BUfoo7yn1YeWLNBeED2rqMGSb1y8Ypv3 > b50d1acc-19eb-455c-8a4b-40263a2d1a9b > a=ssrc:3822531142 mslabel:nqy2BUfoo7yn1YeWLNBeED2rqMGSb1y8Ypv3 > a=ssrc:3822531142 label:b50d1acc-19eb-455c-8a4b-40263a2d1a9b > > > > > second invite (SIPML5 client putting the call on hold) > > INVITE sip:1201@10.24.18.246:5060;transport=WS SIP/2.0 > Via: SIP/2.0/WS > df7jal23ls0d.invalid;branch=z9hG4bK6QhlVcXuVSUNyTewJBtU0KIZzL9mMqCE;rport > From: "sipml_bot"<sip:sipml_bot@10.24.18.246>;tag=W42MEkWLdFTUiIaaw8iy > To: <sip:1201@10.24.18.246>;tag=as269c229d > Contact: > "sipml_bot"<sip:sipml_bot@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr" > Call-ID: 97982d1a-15e5-cd55-9122-07b3cc20cb7d > CSeq: 20908 INVITE > Content-Type: application/sdp > Content-Length: 1840 > Max-Forwards: 70 > User-Agent: IM-client/OMA1.0 sipML5-v1.2014.01.27 > Organization: Doubango Telecom > > v=0 > o=- 3044420497219628500 3 IN IP4 127.0.0.1 > s=Doubango Telecom - chrome > t=0 0 > a=group:BUNDLE audio > a=msid-semantic: WMS nqy2BUfoo7yn1YeWLNBeED2rqMGSb1y8Ypv3 > m=audio 56984 RTP/SAVPF 111 103 104 0 8 106 105 13 126 > c=IN IP4 216.207.245.1 > a=rtcp:56984 IN IP4 216.207.245.1 > a=candidate:474352566 1 udp 2113937151 10.24.16.82 56984 typ host > generation 0 > a=candidate:474352566 2 udp 2113937151 10.24.16.82 56984 typ host > generation 0 > a=candidate:3038348387 1 udp 1845501695 216.207.245.1 56984 typ srflx > raddr 10.24.16.82 rport 56984 generation 0 > a=candidate:3038348387 2 udp 1845501695 216.207.245.1 56984 typ srflx > raddr 10.24.16.82 rport 56984 generation 0 > a=candidate:1388705606 1 tcp 1509957375 10.24.16.82 0 typ host generation > 0 > a=candidate:1388705606 2 tcp 1509957375 10.24.16.82 0 typ host generation > 0 > a=ice-ufrag:LeGzZojC7gQNvl7o > a=ice-pwd:VfqsY2VVKr3xg/mgvdPtcxhp > a=ice-options:google-ice > a=fingerprint:sha-256 > 3A:BA:F5:44:53:CD:99:6C:D1:32:9F:80:53:D4:B5:BA:AE:CF:98:54:71:7F:D6:CB:14:7F:D8:94:30:98:89:62 > a=setup:actpass > a=mid:audio > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=sendonly > a=rtcp-mux > a=crypto:0 AES_CM_128_HMAC_SHA1_32 > inline:ZQQ2MjApfbPlEAmWe52FRkbKUVH4rXt5p5QnpObB > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:JljeDW3NiakHNYn+bu+vje3cD77nNPytUW79J2Vz > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=maxptime:60 > a=ssrc:3822531142 cname:YE/Lm1aHvDjHxwH0 > a=ssrc:3822531142 msid:nqy2BUfoo7yn1YeWLNBeED2rqMGSb1y8Ypv3 > b50d1acc-19eb-455c-8a4b-40263a2d1a9b > a=ssrc:3822531142 mslabel:nqy2BUfoo7yn1YeWLNBeED2rqMGSb1y8Ypv3 > a=ssrc:3822531142 label:b50d1acc-19eb-455c-8a4b-40263a2d1a9b > > > > > third invite (SIMPL5 resumes the call) > > INVITE sip:1201@10.24.18.246:5060;transport=WS SIP/2.0 > Via: SIP/2.0/WS > df7jal23ls0d.invalid;branch=z9hG4bKGAwGP8mOJ7Kd9TuUw4Ac7T1Pc3eVTQgE;rport > From: "sipml_bot"<sip:sipml_bot@10.24.18.246>;tag=W42MEkWLdFTUiIaaw8iy > To: <sip:1201@10.24.18.246>;tag=as269c229d > Contact: > "sipml_bot"<sip:sipml_bot@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr" > Call-ID: 97982d1a-15e5-cd55-9122-07b3cc20cb7d > CSeq: 20909 INVITE > Content-Type: application/sdp > Content-Length: 1838 > Max-Forwards: 70 > User-Agent: IM-client/OMA1.0 sipML5-v1.2014.01.27 > Organization: Doubango Telecom > > v=0 > o=- 11625312205988492 4 IN IP4 127.0.0.1 > s=Doubango Telecom - chrome > t=0 0 > a=group:BUNDLE audio > a=msid-semantic: WMS HpcmWGwoVidQ57oRzreLCtEzVKYzRkuGXCzS > m=audio 43821 RTP/SAVPF 111 103 104 0 8 106 105 13 126 > c=IN IP4 216.207.245.1 > a=rtcp:43821 IN IP4 216.207.245.1 > a=candidate:474352566 1 udp 2113937151 10.24.16.82 43821 typ host > generation 0 > a=candidate:474352566 2 udp 2113937151 10.24.16.82 43821 typ host > generation 0 > a=candidate:3038348387 1 udp 1845501695 216.207.245.1 43821 typ srflx > raddr 10.24.16.82 rport 43821 generation 0 > a=candidate:3038348387 2 udp 1845501695 216.207.245.1 43821 typ srflx > raddr 10.24.16.82 rport 43821 generation 0 > a=candidate:1388705606 1 tcp 1509957375 10.24.16.82 0 typ host generation > 0 > a=candidate:1388705606 2 tcp 1509957375 10.24.16.82 0 typ host generation > 0 > a=ice-ufrag:bBLoaKGIBC2OBJK/ > a=ice-pwd:zJD7a+9Pd54bh3WuLfQEiJRx > a=ice-options:google-ice > a=fingerprint:sha-256 > 3A:BA:F5:44:53:CD:99:6C:D1:32:9F:80:53:D4:B5:BA:AE:CF:98:54:71:7F:D6:CB:14:7F:D8:94:30:98:89:62 > a=setup:actpass > a=mid:audio > a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level > a=sendrecv > a=rtcp-mux > a=crypto:0 AES_CM_128_HMAC_SHA1_32 > inline:4QyVPss3/XPkqz2AcnDioqOwLO+BLQe1T41L8POW > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:dNDtygwacR+DTEppJTmZR4sjLQ/99yIDCBXbZwvJ > a=rtpmap:111 opus/48000/2 > a=fmtp:111 minptime=10 > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=maxptime:60 > a=ssrc:1421740875 cname:vyMP5//bMmdw/mZH > a=ssrc:1421740875 msid:HpcmWGwoVidQ57oRzreLCtEzVKYzRkuGXCzS > d99b2db8-4043-427e-b5b6-6a608f186190 > a=ssrc:1421740875 mslabel:HpcmWGwoVidQ57oRzreLCtEzVKYzRkuGXCzS > a=ssrc:1421740875 label:d99b2db8-4043-427e-b5b6-6a608f186190 > >
Ah ha. They do change on the third INVITE request. We could do this one of two ways: (1) Compare the candidates that were received to what we currently have. If any differ, destroy the ICE session and re-create it. (2) Say damn the torpedoes, full speed ahead - and always destroy the ICE session and re-create it. I'd actually lean to (2), since (1) feels like an optimization you make when you're spending a lot of time destroy/re-creating things. - Matt ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3256/#review10937 ----------------------------------------------------------- On Feb. 24, 2014, 12:36 p.m., Jonathan Rose wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/3256/ > ----------------------------------------------------------- > > (Updated Feb. 24, 2014, 12:36 p.m.) > > > Review request for Asterisk Developers, Joshua Colp, Kevin Harwell, and Matt > Jordan. > > > Bugs: ASTERISK-22911 and ASTERISK-23213 > https://issues.asterisk.org/jira/browse/ASTERISK-22911 > https://issues.asterisk.org/jira/browse/ASTERISK-23213 > > > Repository: Asterisk > > > Description > ------- > > Let me start by saying this is almost certainly not the complete answer to > solving these problems. This patch is simply an alternative to backing out > the patch from r405234 and leaving the existing aborts in place. I've written > a test to make sure the new patch (and likely a later patch which can resolve > these problems with ICE more comprehensively) does not crash Asterisk and > that can be view here: https://reviewboard.asterisk.org/r/3255/ > > In my reproduction of this regression, I noticed a few things. The first was > that when starting a call with ICE from SIPML5 that Asterisk would not be > able to full initialize the ICE session. Instead when creating the candidate > checklist via pj_ice_sess_create_check_list, PJNATH would be unable to > associate the srflx candidates with any host pairs when pruning the > checklist. The SRFLX candidates would have the addresses used internally on > my LAN while the addresses it would be matched up against would mirror those > of my external IP (in other words, they just didn't match by address). The > overall return from pj_ice_sess_create_check_list would be > PJNATH_EICENOHOSTCAND and while the ICE session wouldn't get flagged as > started, I still continued to receive two way audio on both ends in Asterisk > 11.7 > > Asterisk 11.8 however would clear the candidate lists at this point and I > would get one way audio instead. > > The crash in 11.7 from holding was caused because upon holding the call, the > SDP would contain the same list of ICE candidates, so Asterisk would attempt > to start the ICE session again. This time when it entered the create check > list function, the maximum ICE candidates in the session would be exhausted > causing PJNATH to assert and abort (which visibly appears more or less the > same as a crash). This work around patch checks to see if a create checklist > call will cause that value to expand above the maximum and if it would, we > abandon starting ICE back up and we clear the current candidate list on the > ICE session. In my own tests this seemed to work quite well (I had two way > audio, Asterisk didn't terminate suddenly, and I was able to hold and resume > the call while still maintaining two way audio and music on hold for all > expected ends). According to Vytis though, neither this patch nor the > original patch to resolve ASTERISK-22911 by kharwell actually fixed his audio > problems anyway . Kharwell's patch fixed the assertions that were occurring because the candidate list would be cleared on any error including the PJNATH_EICENOHOSTCAND which most of the people in ASTERISK-23213 were most likely experiencing as well. I draw that conclusion because when they used this patch they reported that it fixed their issues without introducing any new crashes. > > At this point, I'm not entirely sure that audio actually should be allowed to > work when building the ICE session check list fails like it has been in this > scenario. But we need to finalize a new Asterisk 11 release soon and we can't > have this apparent regression in when we do so. In the future we need to find > out more comprehensively why pj_ice_sess_create_check_list is failing and how > to fix that. > > > Props to Lott Caskey for providing some improvements to the patch that I > wrote. > > > Diffs > ----- > > /branches/11/res/res_rtp_asterisk.c 408284 > > Diff: https://reviewboard.asterisk.org/r/3256/diff/ > > > Testing > ------- > > Tested calls and holds with a SIPML5 call to a desk phone. Confirmed two way > audio and music on hold. > Requested that the reporter and participants of ASTERISK-23213 test the > patch. Everyone who tested it confirmed that it fixed their problems. > Ran https://reviewboard.asterisk.org/r/3255/ and checked the log files to see > what was happening > > > Thanks, > > Jonathan Rose > >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
