Summary: | ASTERISK-27763: res_pjsip_session: Initial INVITE with audio+fax results in 488 instead of declining stream | ||
Reporter: | Thiago Coutinho (thiagocnet) | Labels: | fax pjsip |
Date Opened: | 2018-03-22 12:19:13 | Date Closed: | 2018-07-09 05:19:09 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Resources/res_pjsip_session |
Versions: | 13.20.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | CentOS Linux release 7.4.1708 (Core) Kernel 3.10.0-693.17.1.el7.x86_64 Asterisk 13.20.0 | Attachments: | |
Description: | 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.
{code:title=pjsip.conf|borderStyle=solid} [voxip] type=registration outbound_auth=voxip server_uri=sip:10.150.129.68 client_uri=sip:4730863277@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 {code} {code:title=pjsip trace|borderStyle=solid} SIP -> Request INVITE sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0 From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56 To:<sip:4731215050@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@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@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0 From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56 To:<sip:4731215050@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@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@10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56 To:<sip:4731215050@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@10.150.129.68;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56 To:<sip:4731215050@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@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0 From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56 To:<sip:4731215050@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@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0 From:<sip:11992567632@10.150.129.68:5060;user=phone>;tag=3c4-45026-7f4a56-9daf24c-7f4a56 To:<sip:4731215050@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 {code} {code:title=sip.conf|borderStyle=solid} [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 {code} {code:title=chan_sip trace|borderStyle=solid} <--- SIP read from UDP:10.150.129.68:5060 ---> INVITE sip:4731215050@10.143.92.98:5060;transport=UDP;user=phone SIP/2.0 From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf To: <sip:4731215050@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@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@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf To: <sip:4731215050@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@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@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf To: <sip:4731215050@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@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@10.143.92.98:5060 SIP/2.0 From: <sip:11987291094@10.150.129.68:5060;user=phone>;tag=-13c4-45026-1c4bf-8b2a32c-1c4bf To: <sip:4731215050@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 <-------------> {code} | ||
Comments: | By: Asterisk Team (asteriskteam) 2018-03-22 12:19:15.974-0500 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]. By: Friendly Automation (friendly-automation) 2018-07-09 05:19:12.396-0500 Change 9352 merged by Joshua Colp: res_pjsip_t38: Decline T.38 stream on failure case. [https://gerrit.asterisk.org/9352|https://gerrit.asterisk.org/9352] By: Friendly Automation (friendly-automation) 2018-07-09 05:34:14.743-0500 Change 9354 merged by Joshua Colp: res_pjsip_t38: Decline T.38 stream on failure case. [https://gerrit.asterisk.org/9354|https://gerrit.asterisk.org/9354] By: Friendly Automation (friendly-automation) 2018-07-09 05:39:20.518-0500 Change 9353 merged by Joshua Colp: res_pjsip_t38: Decline T.38 stream on failure case. [https://gerrit.asterisk.org/9353|https://gerrit.asterisk.org/9353] By: Thiago Coutinho (thiagocnet) 2018-07-09 06:38:29.576-0500 Hi Joshua. With this change the call will be accepted (like in chan_sip) or declined? By: Asterisk Team (asteriskteam) 2018-07-09 06:38:29.845-0500 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: Joshua C. Colp (jcolp) 2018-07-09 06:42:13.375-0500 The audio portion is accepted. The fax portion is declined. |