FS No ACK for 200 OK from GW
Hello all, I am having a situation where 200 OK from a GW arrives at FS and can be seen in FS CLI but no ACK is generated. Here in INVITE and 200 OK packets I am wondering what are the problematic headers I know contact is messy though. Shouldn't it at least generate an ACK ?. PUBLIC-IP:5070 = FS PUBLIC-IP or PUBLIC-IP:5060 = GW(Kamailio) -------------------------- INVITE -------From FS to GW-------------------- INVITE sip:81205999@public.domain.com SIP/2.0 Via: SIP/2.0/UDP PUBLIC-IP:5070;rport;branch=z9hG4bK4rSU85e8rjj6j Max-Forwards: 70 From: "UserA" <sip:16910067@PUBLIC-IP>;tag=BSNjv3XNrFU1m To: <sip:81205999@public.domain.com> Call-ID: 8fe2118a-fe89-1233-2781-069be24d4191 CSeq: 83037069 INVITE Contact: <sip:gw+pcall@PUBLIC-IP:5070;transport=udp;gw=pvcall> User-Agent: FreeSWITCH-mod_sofia/1.4.23+git~20151023T152155Z~9fcf3e0c9a~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 303 X-FS-Support: update_display,send_info Remote-Party-ID: "UserA" <sip:16910067@PUBLIC-IP
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1446729346 1446729347 IN IP4 PUBLIC-IP s=FreeSWITCH c=IN IP4 PUBLIC-IP t=0 0 m=audio 16920 RTP/AVP 102 0 8 9 101 13 a=rtpmap:102 SPEEX/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ---------------------------------- 200 OK -------From GW to FS---------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP PUBLIC-IP:5070;received=PUBLIC-IP;rport=5070;branch=z9hG4bK4rSU85e8rjj6j To: <sip:81205999@public.domain.com>;tag=shv30apv2f From: "UserA" <sip:16910067@PUBLIC-IP>;tag=BSNjv3XNrFU1m Call-ID: 8fe2118a-fe89-1233-2781-069be24d4191 CSeq: 83037069 INVITE Contact: <sip:k471p745@v3n0qcki9ah1.invalid ;alias=182.178.237.31~20760~6;transport=ws> X-Can-Renegotiate: true Supported: ice,replaces,outbound Content-Type: application/sdp Content-Length: 597 v=0 o=- 464531514895581299 2 IN IP4 PUBLIC-IP s=- t=0 0 a=msid-semantic: WMS QT4IzXsqqeNv2uaXsBBrmLBlUeHKFOYKSRMh m=audio 33416 RTP/AVP 0 8 9 101 13 c=IN IP4 PUBLIC-IP a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=rtpmap:13 CN/8000 a=ssrc:4081787962 cname:HWxDJwiTLUiGrbg/ a=ssrc:4081787962 msid:QT4IzXsqqeNv2uaXsBBrmLBlUeHKFOYKSRMh 3f289eda-ce8d-4219-ac04-1147e6a14fdb a=ssrc:4081787962 mslabel:QT4IzXsqqeNv2uaXsBBrmLBlUeHKFOYKSRMh a=ssrc:4081787962 label:3f289eda-ce8d-4219-ac04-1147e6a14fdb a=sendrecv a=rtcp:33417 -- Regards Salman
Try turning up logs so you get more info and get a complete trace sofia global siptrace on console loglevel debug sofia loglevel all 9 sofia tracelevel alert Also post things like this to https://freeswitch.org/jira not to the list. On Thu, Nov 5, 2015 at 12:40 PM, Salman <msalman212@gmail.com> wrote:
Hello all,
I am having a situation where 200 OK from a GW arrives at FS and can be seen in FS CLI but no ACK is generated. Here in INVITE and 200 OK packets I am wondering what are the problematic headers I know contact is messy though. Shouldn't it at least generate an ACK ?.
PUBLIC-IP:5070 = FS PUBLIC-IP or PUBLIC-IP:5060 = GW(Kamailio)
-------------------------- INVITE -------From FS to GW-------------------- INVITE sip:81205999@public.domain.com SIP/2.0 Via: SIP/2.0/UDP PUBLIC-IP:5070;rport;branch=z9hG4bK4rSU85e8rjj6j Max-Forwards: 70 From: "UserA" <sip:16910067@PUBLIC-IP>;tag=BSNjv3XNrFU1m To: <sip:81205999@public.domain.com> Call-ID: 8fe2118a-fe89-1233-2781-069be24d4191 CSeq: 83037069 INVITE Contact: <sip:gw+pcall@PUBLIC-IP:5070;transport=udp;gw=pvcall> User-Agent: FreeSWITCH-mod_sofia/1.4.23+git~20151023T152155Z~9fcf3e0c9a~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 303 X-FS-Support: update_display,send_info Remote-Party-ID: "UserA" <sip:16910067@PUBLIC-IP
;party=calling;screen=yes;privacy=off
v=0 o=FreeSWITCH 1446729346 1446729347 IN IP4 PUBLIC-IP s=FreeSWITCH c=IN IP4 PUBLIC-IP t=0 0 m=audio 16920 RTP/AVP 102 0 8 9 101 13 a=rtpmap:102 SPEEX/16000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20
---------------------------------- 200 OK -------From GW to FS---------------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP PUBLIC-IP:5070;received=PUBLIC-IP;rport=5070;branch=z9hG4bK4rSU85e8rjj6j To: <sip:81205999@public.domain.com>;tag=shv30apv2f From: "UserA" <sip:16910067@PUBLIC-IP>;tag=BSNjv3XNrFU1m Call-ID: 8fe2118a-fe89-1233-2781-069be24d4191 CSeq: 83037069 INVITE Contact: <sip:k471p745@v3n0qcki9ah1.invalid ;alias=182.178.237.31~20760~6;transport=ws> X-Can-Renegotiate: true Supported: ice,replaces,outbound Content-Type: application/sdp Content-Length: 597
v=0 o=- 464531514895581299 2 IN IP4 PUBLIC-IP s=- t=0 0 a=msid-semantic: WMS QT4IzXsqqeNv2uaXsBBrmLBlUeHKFOYKSRMh m=audio 33416 RTP/AVP 0 8 9 101 13 c=IN IP4 PUBLIC-IP a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=rtpmap:13 CN/8000 a=ssrc:4081787962 cname:HWxDJwiTLUiGrbg/ a=ssrc:4081787962 msid:QT4IzXsqqeNv2uaXsBBrmLBlUeHKFOYKSRMh 3f289eda-ce8d-4219-ac04-1147e6a14fdb a=ssrc:4081787962 mslabel:QT4IzXsqqeNv2uaXsBBrmLBlUeHKFOYKSRMh a=ssrc:4081787962 label:3f289eda-ce8d-4219-ac04-1147e6a14fdb a=sendrecv a=rtcp:33417
-- Regards Salman
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participants (2)
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Anthony Minessale
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Salman