[asterisk-bugs] [JIRA] (ASTERISK-27763) PJSIP doesn't accept a call when INVITE contains T.38 stream, chan_sip does
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Thu Mar 22 12:19:40 CDT 2018
[ https://issues.asterisk.org/jira/browse/ASTERISK-27763?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=242779#comment-242779 ]
Asterisk Team commented on ASTERISK-27763:
------------------------------------------
Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.
A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.
Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
> PJSIP doesn't accept a call when INVITE contains T.38 stream, chan_sip does
> ---------------------------------------------------------------------------
>
> Key: ASTERISK-27763
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-27763
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: pjproject/pjsip
> Affects Versions: 13.20.0
> Environment: CentOS Linux release 7.4.1708 (Core)
> Kernel 3.10.0-693.17.1.el7.x86_64
> Asterisk 13.20.0
> Reporter: Thiago Coutinho
> Labels: fax, pjsip
>
> Some providers send T.38 streams along with the call (I don't know why) causing PJSIP to reject the call. chan_sip on the other hand accepts the call normally.
> pjsip.conf:
> ```
> [voxip]
> type=registration
> outbound_auth=voxip
> server_uri=sip:10.150.129.68
> client_uri=sip:4730863277 at 10.150.129.68
> auth_rejection_permanent=no
> [voxip]
> type=auth
> auth_type=userpass
> username=4730863277
> password=4730863277
> [voxip]
> type=aor
> contact=sip:10.150.129.68
> qualify_frequency=60
> [voxip]
> type=endpoint
> context=from-pstn
> allow=!all,g729,alaw
> ;auth=voxip
> outbound_auth=voxip
> aors=voxip
> rtp_symmetric=yes
> force_rport=yes
> rewrite_contact=yes
> from_user=4730863277
> from_domain=10.150.129.68
> t38_udptl=yes
> t38_udptl_ec=redundancy
> fax_detect=no
> t38_udptl_nat=yes
> [voxip]
> type=identify
> endpoint=voxip
> match=10.150.129.68
> ```
> pjsip trace:
> ```
> SIP ->
> Request
> INVITE sip:4731215050 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
> From:<sip:11992567632 at 10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
> To:<sip:4731215050 at 10.143.92.98:5060;user=phone>
> Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
> CSeq: 1 INVITE
> User-agent:CS2000_NGSS/9.0
> P-Asserted-Identity:<sip:11992567632 at 10.150.129.68;user=phone>
> Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
> Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
> Max-Forwards:140
> Contact:<sip:10.150.129.68:5060;transport=UDP>
> Supported:100rel,resource-priority
> Content-Type: application/sdp
> Content-Length:420
> SDP ->
> Version = 0.
> Owner = PVG 1521481511010 1521481511010 IN IP4 10.152.205.107.
> Session Name = -.
> Phone Address = +1 6135555555.
> Connection = IN IP4 10.152.205.107.
> Time = 0 0.
> Media Name = audio 56534 RTP/AVP 18 8 101.
> Media Attribute = rtpmap:101 telephone-event/8000.
> Media Attribute = a=fmtp:101 0-15.
> Media Attribute = a=ptime:20.
> Media Attribute = a=fmtp:18 annexb=no.
> Media Attribute = m=image 64726 udptl t38.
> Media Attribute = a=T38FaxVersion:0.
> Media Attribute = a=T38FaxMaxBuffer:1100.
> Media Attribute = a=T38FaxMaxDatagram:612.
> Media Attribute = a=T38MaxBitRate:14400.
> Media Attribute = a=T38FaxRateManagement:transferredTCF.
> Media Attribute = a=T38FaxUdpEC:t38UDPRedundancy.
> SIP ->
> Request
> INVITE sip:4731215050 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
> From:<sip:11992567632 at 10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
> To:<sip:4731215050 at 10.143.92.98:5060;user=phone>
> Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
> CSeq: 1 INVITE
> User-agent:CS2000_NGSS/9.0
> P-Asserted-Identity:<sip:11992567632 at 10.150.129.68;user=phone>
> Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
> Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
> Max-Forwards:140
> Contact:<sip:10.150.129.68:5060;transport=UDP>
> Supported:100rel,resource-priority
> Content-Type: application/sdp
> Content-Length:420
> SDP ->
> Version = 0.
> Owner = PVG 1521481511010 1521481511010 IN IP4 10.152.205.107.
> Session Name = -.
> Phone Address = +1 6135555555.
> Connection = IN IP4 10.152.205.107.
> Time = 0 0.
> Media Name = audio 56534 RTP/AVP 18 8 101.
> Media Attribute = rtpmap:101 telephone-event/8000.
> Media Attribute = a=fmtp:101 0-15.
> Media Attribute = a=ptime:20.
> Media Attribute = a=fmtp:18 annexb=no.
> Media Attribute = m=image 64726 udptl t38.
> Media Attribute = a=T38FaxVersion:0.
> Media Attribute = a=T38FaxMaxBuffer:1100.
> Media Attribute = a=T38FaxMaxDatagram:612.
> Media Attribute = a=T38MaxBitRate:14400.
> Media Attribute = a=T38FaxRateManagement:transferredTCF.
> Media Attribute = a=T38FaxUdpEC:t38UDPRedundancy.
> SIP <-
> Response
> SIP/2.0 488 Not Acceptable Here
> Via:SIP/2.0/UDP SOO2CS2K:5060;rport=5060;maddr=10.150.129.68;received=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
> Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
> From:<sip:11992567632 at 10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
> To:<sip:4731215050 at 10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
> CSeq: 1 INVITE
> Server:Asterisk PBX certified/13.13-cert7
> Content-Length:0
> SIP <-
> Response
> SIP/2.0 488 Not Acceptable Here
> Via:SIP/2.0/UDP SOO2CS2K:5060;rport=5060;maddr=10.150.129.68;received=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
> Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
> From:<sip:11992567632 at 10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
> To:<sip:4731215050 at 10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
> CSeq: 1 INVITE
> Server:Asterisk PBX certified/13.13-cert7
> Content-Length:0
> SIP ->
> Request
> ACK sip:4731215050 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
> From:<sip:11992567632 at 10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
> To:<sip:4731215050 at 10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
> Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
> CSeq: 1 ACK
> User-agent:CS2000_NGSS/9.0
> Max-Forwards:70
> Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
> Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
> Contact:<sip:10.150.129.68:5060;transport=UDP>
> Supported:100rel,resource-priority
> Content-Length:0
> SIP <-
> Request
> ACK sip:4731215050 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
> From:<sip:11992567632 at 10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56
> To:<sip:4731215050 at 10.143.92.98;user=phone>;tag=9f804ff5-215a-4c91-a93e-52e4689f866c
> Call-ID:eb76d2304481960a13c47f4a56f13a6220a1b439c2a972a640-0008-7943
> CSeq: 1 ACK
> User-agent:CS2000_NGSS/9.0
> Max-Forwards:70
> Allow:ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
> Via:SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-7f4a56-f13a6220-5c9ab8e5
> Contact:<sip:10.150.129.68:5060;transport=UDP>
> Supported:100rel,resource-priority
> Content-Length:0
> ```
> sip.conf:
> ```
> [voxip]
> type=peer
> defaultuser=4730863277
> secret=4730863277
> fromuser=4730863277
> fromdomain=gvt.com.br
> domain=gvt.com.br
> host=10.150.129.68
> context=from-pstn
> dtmfmode=rfc2833
> insecure=port,invite
> qualify=yes
> canreinvite=no
> disallow=all
> allow=alaw
> nat=no
> port=5060
> ignoresdpversion=yes
> busydetect=yes
> busycount=3
> t38pt_udptl=yes
> ```
> chan_sip trace:
> ```
> <--- SIP read from UDP:10.150.129.68:5060 --->
> INVITE sip:4731215050 at 10.143.92.98:5060;transport=UDP;user=phone SIP/2.0
> From: <sip:11987291094 at 10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
> To: <sip:4731215050 at 10.143.92.98:5060;user=phone>
> Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
> CSeq: 1 INVITE
> User-agent: CS2000_NGSS/9.0
> P-Asserted-Identity: <sip:11987291094 at 10.150.129.68;user=phone>
> Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
> Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c5b-43954845
> Max-Forwards: 140
> Contact: <sip:10.150.129.68:5060;transport=UDP>
> Supported: 100rel,resource-priority
> Content-Type: application/sdp
> Content-Length: 418
> v=0
> o=PVG 1521732832740 1521732832740 IN IP4 10.152.204.43
> s=-
> p=+1 6135555555
> c=IN IP4 10.152.204.43
> t=0 0
> m=audio 49330 RTP/AVP 18 8 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=fmtp:18 annexb=no
> m=image 57522 udptl t38
> a=T38FaxVersion:0
> a=T38FaxMaxBuffer:1100
> a=T38FaxMaxDatagram:612
> a=T38MaxBitRate:14400
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxUdpEC:t38UDPRedundancy
> <------------->
> --- (14 headers 18 lines) ---
> Sending to 10.150.129.68:5060 (NAT)
> Sending to 10.150.129.68:5060 (NAT)
> Using INVITE request as basis request - 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
> Found peer 'VOXIP_GVT' for '11987291094' from 10.150.129.68:5060
> == Using SIP RTP CoS mark 5
> Found RTP audio format 18
> Found RTP audio format 8
> Found RTP audio format 101
> Found audio description format telephone-event for ID 101
> == Using UDPTL CoS mark 5 [107/1736]
> Got T.38 offer in SDP in dialog 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
> Capabilities: us - (alaw), peer - audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port 10.152.204.43:49330
> Looking for 4731215050 in from-pstn (domain 10.143.92.98)
> sip_route_dump: route/path hop: <sip:10.150.129.68:5060;transport=UDP>
> <--- Transmitting (no NAT) to 10.150.129.68:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c5b-43954845;received=10.150.129.68
> From: <sip:11987291094 at 10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
> To: <sip:4731215050 at 10.143.92.98:5060;user=phone>
> Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.20.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:4731215050 at 10.143.92.98:5060>
> Content-Length: 0
> <------------>
> Audio is at 14648
> Adding codec alaw to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> <--- Reliably Transmitting (no NAT) to 10.150.129.68:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c5b-43954845;received=10.150.129.68
> From: <sip:11987291094 at 10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
> To: <sip:4731215050 at 10.143.92.98:5060;user=phone>;tag=as2030a2ce
> Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.20.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:4731215050 at 10.143.92.98:5060>
> Content-Type: application/sdp
> Content-Length: 259
> v=0
> o=root 413021425 413021425 IN IP4 10.143.92.98 [61/1736]
> s=Asterisk PBX 13.20.0
> c=IN IP4 10.143.92.98
> t=0 0
> m=audio 14648 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=maxptime:150
> a=sendrecv
> m=image 0 udptl t38
> <------------>
> <--- SIP read from UDP:10.150.129.68:5060 --->
> ACK sip:4731215050 at 10.143.92.98:5060 SIP/2.0
> From: <sip:11987291094 at 10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf
> To: <sip:4731215050 at 10.143.92.98:5060;user=phone>;tag=as2030a2ce
> Call-ID: 7f9af3004481960a13c41c4bf6e88c5b4097a7dd504d8f40-0058-4883
> CSeq: 1 ACK
> User-agent: CS2000_NGSS/9.0
> Max-Forwards: 70
> Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
> Via: SIP/2.0/UDP SOO2CS2K:5060;maddr=10.150.129.68;branch=z9hG4bK-1c4bf-6e88c76-6ec0f83c
> Contact: <sip:10.150.129.68:5060;transport=UDP>
> Supported: 100rel,resource-priority
> Content-Length: 0
> <------------->
> ```
--
This message was sent by Atlassian JIRA
(v6.2#6252)
More information about the asterisk-bugs
mailing list