[asterisk-bugs] [JIRA] (ASTERISK-29030) res_rtp_asterisk: Additional RTP-frame (with wrong SSRC) gets inserted when switching from progress to established
Sean Bright (JIRA)
noreply at issues.asterisk.org
Wed Apr 21 15:55:10 CDT 2021
[ https://issues.asterisk.org/jira/browse/ASTERISK-29030?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Sean Bright updated ASTERISK-29030:
-----------------------------------
Assignee: Matthias Hensler (was: Sean Bright)
Status: Waiting for Feedback (was: Open)
> res_rtp_asterisk: Additional RTP-frame (with wrong SSRC) gets inserted when switching from progress to established
> ------------------------------------------------------------------------------------------------------------------
>
> Key: ASTERISK-29030
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-29030
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General, Resources/res_rtp_asterisk
> Affects Versions: 16.10.0, 16.11.0, 16.12.0
> Environment: CentOS 7
> Reporter: Matthias Hensler
> Assignee: Matthias Hensler
> Labels: patch
> Attachments: 0001-WIP-res_rtp_asterisk.c-Don-t-assuming-timestamp-0-is.patch, 60350068.pcap, 60350082.pcap, 60408587.pcap, 64008592.pcap, debug_all2.txt
>
>
> Hi there,
> we have a strange problem in which a broken RTP-frame (old sequence-number but
> new ssrc) is sent in the moment a call gets established after its progress
> phase.
> In that scenario we are receiving calls from our upstream provider (via
> TELES-SBC) and forwarding them to a customer. Both calllegs are provided via
> chan_sip and use G711a as voicecodec. A ringtone is provided via
> progressinband currently (I want to switch that off, as this is at least
> part of the problem).
> Before I add the details I have to emphasize, that only calls to one specific
> customer show the problem, and no other calls which are distributed in exact the
> same way. So far, after checking numerous traces, I have no clue what so ever
> about which difference in the setup for this one customer is, that leads to
> this issue.
> Via upstream this bug is causing the Teles to drop all audioframes after
> receiving that broken RTP frame.
> It might (and I cannot really confirm this at this moment) that the bug was
> introduced with Asterisk 16.10.0, as the first problem report was shortly after we
> updated from 16.9.0. So far we already updated to 16.11.0 and 16.12.0 which
> all show the same issue (however the problem is so obscure that it took us
> really long to figure the rootcause for the customers problem).
> As such I can say with confidence that the problem is present in versions
> 16.10.0, 16.11.0 and 16.12.0, while 16.9.0 could be ok. To make matters
> complicated in pointing to a specific version is, that the customer also did
> several updates of his software in the meantime, so it might be difficult to
> rollback to the specific version in which everything was ok.
> I am not sure on how to proceed here, to find the point in where Asterisk gets
> mixed up in the RTP flow, so I am hoping you have either a clue where the
> issue might come from or on how to debug (and of course fix it).
> This Asterisk installation processes several 10000 calls a day, with >200 calls
> in parallel, but as said only one client triggers that issue, while all other
> clients are fine.
> OK, up to the setup and configuration (I "XXX"ed some values):
> Our "sip.conf" looks like this:
> {noformat}
> [general]
> context=inbound_mapping
> allowoverlap=no
> realm=XXX
> bindport=5060
> language=de
> canreinvite=no
> alwaysauthreject=yes
> allowguest=no
> bindaddr=::
> tcpenable=yes
> tcpbindaddr=::
> tlsenable=yes
> tlsbindaddr=::
> tlscertfile=/etc/asterisk/certificate.pem
> tlscipher=HIGH:!aNULL:!eNULL:!EXPORT:!DES:!RC4:!MD5:!PSK
> tlsclientmethod=tlsv1
> transport=udp,tls
> srvlookup=no
> allowsubscribe=no
> disallow=all
> allow=g722
> allow=alaw
> t38pt_udptl=yes,redundancy,maxdatagram=400
> progressinband=yes
> prematuremedia=no
> sendrpid=yes
> trustrpid=no
> send_diversion=no
> allowtransfer=no
> tos_sip=cs3
> tos_audio=ef
> [our_inbound]
> type=peer
> host=XXX.XXX.XXX.XXX
> qualify=yes
> context=inbound_provider
> disallow=all
> allow=g722
> allow=alaw
> [iads](!)
> type=friend
> context=iads_unregistered
> dtmfmode=auto
> host=dynamic
> amaflags=documentation
> canreinvite=no
> ignoresdpversion=yes
> [the_customer](iads)
> type=friend
> accountcode=0049XXXX
> setvar=NOTRUF_UUI=002DXXXXXXX
> secret=XXXXX
> context=iads_registered
> acl=public
> {noformat}
> The interesting part of our "extensions.conf" looks like this:
> {noformat}
> [inbound_provider]
> exten => _+X.,1,Goto(00${EXTEN:1},1)
> exten => _X.,1,Noop(Inboundcall XXX)
> exten => _X.,n,GotoIf($[${DIALPLAN_EXISTS(routing_intern,${EXTEN},1)} = 1]?DoRouting:)
> exten => _X.,n,Noop(Zielnummer ${EXTEN} existiert nicht)
> exten => _X.,n,Hangup(1)
> exten => _X.,n(DoRouting),Goto(routing_intern,${EXTEN},1)
> [routing_intern]
> include => inbound_mapping
> [inbound_mapping]
> exten => _0049XXX!,1,Set(Peerstring="SIP/the_customer")
> exten => _0049XXX!,n(DoMapping),Gosub(gosub-do_mapping,${EXTEN},1(${Peerstring},3,0049,,0049XXX,XXX,XXX,XXX,XX))
> [gosub-do_mapping]
> exten => _00Z.,1,Noop(Mapping)
> exten => _00Z.,n,Set(DIAL_ARGS=)
> exten => _00Z.,n,Set(CallTimeout=180)
> exten => _00Z.,n,Dial(${ARG1},${CallTimeout},${DIAL_ARGS})
> {noformat}
> So in short: an inbound call from "our_inbound" is signaled to
> "Dial(SIP/our_customer,180,)". All stuff inbetween is just for checking
> channellimits, billing, redirections, etc. and should not be important for
> this issue.
> In the following I will present the SIP-traces and RTP-packets of a broken
> call. In that I switched the ip-address of "our_inbound" to 1.1.1.1, the
> ip-address of the Asteriskserver to 2.2.2.2 and the ip-address of the client
> "the_customer" to 3.3.3.3. (For our reference the pcap-trace for LAG 1 which
> is 1.1.1.1 <-> 2.2.2.2 has the id 60155926 and the trace for LAG 2 which is
> 2.2.2.2 <-> 3.3.3.3 has the id 60155922). In case of "our_inbound" the
> RTP-media is terminated at a different ip for which I will use 1.1.1.2.
> {noformat}
> -----------------------------------------------------------------------------
> At 0.000000 the call is initiated by "our_inbound":
> -----------------------------------------------------------------------------
> INVITE sip:+49XXX at customer.XXX.net;transport=udp;user=phone SIP/2.0
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK00E0F515158AAD9FCE106BF410B7
> From: "+49XXX" <sip:+49XXX at mgc-cluster.XXX.net;user=phone>;tag=0UUHS0000030000E1D0101Cu045FV8G07E1469
> To: "+49XXX" <sip:+49XXX at customer.XXX.net;user=phone>
> Call-ID: 8415e00015f5-5f36402b-2d18768f-23c282b0-3916f46-01
> CSeq: 163 INVITE
> Contact: <sip:sgc_c at 1.1.1.1:5060;transport=udp>
> Allow-Events: refer
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
> Content-Type: application/sdp
> Max-Forwards: 70
> P-Asserted-Identity: <sip:+49XXX at mgc-cluster.XXX.net;user=phone>
> P-Preferred-Identity: <sip:+49XXX at mgc-cluster.XXX.net>
> Privacy: none
> Supported: timer, replaces, histinfo, 100rel
> User-Agent: TELES-SBC
> Content-Length: 229
> v=0
> o=- 2285767812167987084 1 IN IP4 1.1.1.1
> s=TELES-SBC
> c=IN IP4 1.1.1.2
> t=0 0
> m=audio 11940 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=silenceSupp:off - - - -
> -----------------------------------------------------------------------------
> At 0.001351 Asterisk is replying with Trying:
> -----------------------------------------------------------------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK00E0F515158AAD9FCE106BF410B7;received=1.1.1.1
> From: "+49XXX" <sip:+49XXX at mgc-cluster.XXX.net;user=phone>;tag=0UUHS0000030000E1D0101Cu045FV8G07E1469
> To: "+49XXX" <sip:+49XXX at customer.XXX.net;user=phone>
> Call-ID: 8415e00015f5-5f36402b-2d18768f-23c282b0-3916f46-01
> CSeq: 163 INVITE
> Server: Asterisk PBX 16.12.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:+49XXX at 2.2.2.2:5060>
> Content-Length: 0
> -----------------------------------------------------------------------------
> In parallel Asterisk now signales "our_customer":
> -----------------------------------------------------------------------------
> INVITE sip:+49XXX at 3.3.3.3 SIP/2.0
> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK7d70de6a
> Max-Forwards: 70
> From: <sip:+49XXX at 2.2.2.2>;tag=as437a6f2f
> To: <sip:+49XXX at 3.3.3.3>
> Contact: <sip:+49XXX at 2.2.2.2:5060>
> Call-ID: 46f3325b1867f4985b09200e05ff2519 at 2.2.2.2:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 16.12.0
> Date: Fri, 14 Aug 2020 07:41:31 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Remote-Party-ID: "+49XXX" <sip:+49XXX at 2.2.2.2>;party=calling;privacy=off;screen=no
> Content-Type: application/sdp
> Content-Length: 278
> v=0
> o=root 1849926336 1849926336 IN IP4 2.2.2.2
> s=Asterisk PBX 16.12.0
> c=IN IP4 2.2.2.2
> t=0 0
> m=audio 22504 RTP/AVP 8 9 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> -----------------------------------------------------------------------------
> First the customer replies with Trying:
> -----------------------------------------------------------------------------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK7d70de6a
> To: <sip:+49XXX at 3.3.3.3>
> From: <sip:+49XXX at 2.2.2.2>;tag=as437a6f2f
> Call-ID: 46f3325b1867f4985b09200e05ff2519 at 2.2.2.2:5060
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE
> User-Agent: Swyx LinkMgr/12.11 (SW.Rel12.10_20200423.1)
> Content-Length: 0
> -----------------------------------------------------------------------------
> And then (after around 0.4s) with Ringing:
> -----------------------------------------------------------------------------
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK7d70de6a
> Contact: <sip:+49XXX at 3.3.3.3:65002;transport=udp>
> To: "XXX, XXX"<sip:+49XXX at 3.3.3.3>;tag=a236c740
> From: <sip:+49XXX at 2.2.2.2>;tag=as437a6f2f
> Call-ID: 46f3325b1867f4985b09200e05ff2519 at 2.2.2.2:5060
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE
> User-Agent: Swyx LinkMgr/12.11 (SW.Rel12.10_20200423.1)
> Content-Length: 0
> -----------------------------------------------------------------------------
> With that Asterisk starts (at 0.426196) ringing to "our_inbound":
> -----------------------------------------------------------------------------
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK00E0F515158AAD9FCE106BF410B7;received=1.1.1.1
> From: "+49XXX" <sip:+49XXX at mgc-cluster.XXX.net;user=phone>;tag=0UUHS0000030000E1D0101Cu045FV8G07E1469
> To: "+49XXX" <sip:+49XXX at customer.XXX.net;user=phone>;tag=as32a03bca
> Call-ID: 8415e00015f5-5f36402b-2d18768f-23c282b0-3916f46-01
> CSeq: 163 INVITE
> Server: Asterisk PBX 16.12.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:+49XXX at 2.2.2.2:5060>
> Content-Length: 0
> -----------------------------------------------------------------------------
> And then (because of "inbandprogress=yes") Asterisk sends progress and starts
> sending RTP:
> -----------------------------------------------------------------------------
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK00E0F515158AAD9FCE106BF410B7;received=1.1.1.1
> From: "+49XXX" <sip:+49XXX at mgc-cluster.XXX.net;user=phone>;tag=0UUHS0000030000E1D0101Cu045FV8G07E1469
> To: "+49XXX" <sip:+49XXX at customer.XXX.net;user=phone>;tag=as32a03bca
> Call-ID: 8415e00015f5-5f36402b-2d18768f-23c282b0-3916f46-01
> CSeq: 163 INVITE
> Server: Asterisk PBX 16.12.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:+49XXX at 1.1.1.1:5060>
> Content-Type: application/sdp
> Require: timer
> Content-Length: 252
> v=0
> o=root 829526440 829526440 IN IP4 1.1.1.1
> s=Asterisk PBX 16.12.0
> c=IN IP4 1.1.1.1
> t=0 0
> m=audio 46586 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> -----------------------------------------------------------------------------
> For the next 3.2s Asterisk is sending RTP frames to 1.1.1.2 containing a
> ringtone. In total 161 packets are sent with the SSRC 783be076.
> The flow is 2.2.2.2:46586 -> 1.1.1.2:11940
> -----------------------------------------------------------------------------
> PT=ITU-T G.711 PCMA, SSRC=0x783BE076, Seq=27297, Time=160, Mark
> PT=ITU-T G.711 PCMA, SSRC=0x783BE076, Seq=27298, Time=320
> PT=ITU-T G.711 PCMA, SSRC=0x783BE076, Seq=27299, Time=480
> [...]
> PT=ITU-T G.711 PCMA, SSRC=0x783BE076, Seq=27455, Time=25440
> PT=ITU-T G.711 PCMA, SSRC=0x783BE076, Seq=27456, Time=25600
> PT=ITU-T G.711 PCMA, SSRC=0x783BE076, Seq=27457, Time=25760
> -----------------------------------------------------------------------------
> Now "our_customer" answers the call:
> -----------------------------------------------------------------------------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK7d70de6a
> Contact: <sip:+49XXX at 2.2.2.2:65002;transport=udp>
> To: "XXX, XXX"<sip:+49XXX at 3.3.3.3>;tag=a236c740
> From: <sip:+49XXX at 2.2.2.2>;tag=as437a6f2f
> Call-ID: 46f3325b1867f4985b09200e05ff2519 at 2.2.2.2:5060
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE, UPDATE
> Content-Type: application/sdp
> User-Agent: Swyx LinkMgr/12.11 (SW.Rel12.10_20200423.1)
> Content-Length: 208
> v=0
> o=- 3330958455 1 IN IP4 3.3.3.3
> s=Swyx LinkMgr
> c=IN IP4 3.3.3.3
> t=0 0
> m=audio 55542 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> -----------------------------------------------------------------------------
> Asterisk acks:
> -----------------------------------------------------------------------------
> ACK sip:+49XXX at 3.3.3.3:65002;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK0ea35c4b
> Max-Forwards: 70
> From: <sip:+49XXX at 2.2.2.2>;tag=as437a6f2f
> To: <sip:+49XXX at 3.3.3.3>;tag=a236c740
> Contact: <sip:+49XXX at 2.2.2.2:5060>
> Call-ID: 46f3325b1867f4985b09200e05ff2519 at 2.2.2.2:5060
> CSeq: 102 ACK
> User-Agent: Asterisk PBX 16.12.0
> Content-Length: 0
> {noformat}
> -----------------------------------------------------------------------------
> RTP between Asterisk and "our_customer" is exchanged.
> Asterisk sends 311 packets with SSRC 7f5a528c from 2.2.2.2:22504 to
> 3.3.3.3:55542, while "our_customer" sends 317 packets with SSRC 6b48138f from
> 3.3.3.3:55542 to 2.2.2.2:22504:
> -----------------------------------------------------------------------------
> {noformat}
> 3.687479 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14272, Time=0, Mark
> 3.687489 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14273, Time=160
> 3.707447 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14274, Time=320
> 3.726456 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14275, Time=480
> 3.746436 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14276, Time=640
> 3.764300 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=140, Time=3320682905, Mark
> 3.766431 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14277, Time=800
> 3.774281 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=141, Time=3320683065
> 3.786686 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14278, Time=960
> 3.794455 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=142, Time=3320683225
> [...]
> 9.920692 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14587, Time=50400
> 9.934279 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=449, Time=3320732345
> 9.41389 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14588, Time=50560
> 9.964234 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=450, Time=3320732505
> {noformat}
> -----------------------------------------------------------------------------
> Meanwhile Asterisk establishes (at 3.660704) the call with "our_inbound"
> -----------------------------------------------------------------------------
> {noformat}
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK00E0F515158AAD9FCE106BF410B7;received=1.1.1.1
> From: "+49XXX" <sip:+49XXX at mgc-cluster.XXX.net;user=phone>;tag=0UUHS0000030000E1D0101Cu045FV8G07E1469
> To: "+49XXX" <sip:+49XXX at customer.XXX.net;user=phone>;tag=as32a03bca
> Call-ID: 8415e00015f5-5f36402b-2d18768f-23c282b0-3916f46-01
> CSeq: 163 INVITE
> Server: Asterisk PBX 16.12.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:+49XXX at 2.2.2.2:5060>
> Content-Type: application/sdp
> Require: timer
> Content-Length: 252
> v=0
> o=root 829526440 829526440 IN IP4 2.2.2.2
> s=Asterisk PBX 16.12.0
> c=IN IP4 2.2.2.2
> t=0 0
> m=audio 46586 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> -----------------------------------------------------------------------------
> "our_inbound" acks
> -----------------------------------------------------------------------------
> ACK sip:+49XXX at 2.2.2.2:5060 SIP/2.0
> Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK00E0F515158AAD9FCF7F6278F296
> From: "+49XXX" <sip:+49XXX at mgc-cluster.XXX.net;user=phone>;tag=0UUHS0000030000E1D0101Cu045FV8G07E1469
> To: "+49XXX" <sip:+49XXX at customer.XXX.net;user=phone>;tag=as32a03bca
> Call-ID: 8415e00015f5-5f36402b-2d18768f-23c282b0-3916f46-01
> CSeq: 163 ACK
> Contact: <sip:+49XXX at 1.1.1.1:5060;transport=udp>
> Max-Forwards: 70
> Content-Length: 0
> {noformat}
> -----------------------------------------------------------------------------
> And now the fun part with the real issues begins.
> "our_inbound" sends 313 RTP-packets with SSRC 7f5a528c from 1.1.1.2:11940 to
> 2.2.2.2:46586. Asterisk switches its SSRC to 6b48138f sending 316 packets from
> 2.2.2.2:46586 to 1.1.1.2:11940.
> However, just after starting Asterisk includes another RTP-frame with the new
> SSRC 6b48138f but with the old(!) sequence number. At this moment RTP breaks
> on the Teles at "our_inbound".
> -----------------------------------------------------------------------------
> {noformat}
> 3.695997 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14273, Time=160, Mark
> 3.696054 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=27458, Time=0 (<---- !)
> 3.715943 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14274, Time=320
> 3.734968 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14275, Time=480
> 3.754930 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14276, Time=640
> 3.762733 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=140, Time=3320682905
> 3.774945 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14277, Time=800
> 3.782714 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=141, Time=3320683065
> 3.795174 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14278, Time=960
> 3.802902 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=142, Time=3320683225
> [...]
> 9.942691 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=449, Time=3320732345
> 9.949898 PT=ITU-T G.711 PCMA, SSRC=0x6B48138F, Seq=14588, Time=50560
> 9.962661 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=450, Time=3320732505
> 9.983132 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=451, Time=3320732655
> 10.007195 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=452, Time=3320732825
> 10.022785 PT=ITU-T G.711 PCMA, SSRC=0x7F5A528C, Seq=453, Time=3320732985
> {noformat}
> -----------------------------------------------------------------------------
> I am speculating, but something causes Asterisk to send a spurious RTP-frame
> which seems to belong to the inband-progress signal (as it uses the old
> sequencenumbers), but the RTP-stack does not longer know the old stream and
> thus sending it with the new SSRC-identifier and a timestamp of 0.
> Still, that issue seems to only affect calls which a routed to "our_customer".
> As to why I am at total loss, as I checked a lot of calls to other clients,
> which all look the same (Asterisk creating an inband ringtone and switching to
> a new SSRC when the call established). All calls to the other customers make a
> perfect switch on the leg to "our_inbound", only on calls to "our_customer"
> show that spurious RTP-frame.
> As always I am happy to provide any additional infos or details needed.
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