transport=tls is not working while sending SIP INVITE from pjsip to sip proxy

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Hi Shubham,

The ID sets the contact address, I think you also need to specify 
";transport=tls" in the SIP invite address when making the call.

Regards,

Bill

On 12/11/2015 3:06 AM, shubham verma wrote:
> I am using TLS transport in PJSIP AND HAS SET FOLLOWING PARAMETERS IN CFG FILE
>
> --id sip:911126592065 at 192.168.103.175:5061;transport=tls
> --registrar sip:192.168.103.175;transport=tls
>
>
> The register request is successfully sent using transport=tls  option
> in registrar but on using transport=tls in id(sip uri), pjsip sends
> Invite on UDP rather than TLS
>
> phone got registered to server as follows :
>
> REGISTER sips:192.168.103.175;transport=tls SIP/2.0
> Via: SIP/2.0/TLS
> 192.168.1.186:41924;rport;branch=z9hG4bKPjDhTOMSWiHtXVLW5tKMonLGlaSVtHUoEX;alias
> Max-Forwards: 70
> From: <sip:911126592065@192.168.103.175>;tag=9R61SCQehk-ZLda4TP-QVboszhEcyqU0
> To: <sip:911126592065 at 192.168.103.175>
> Call-ID: BZqPf3rFRXFueLb.LcxrL0hWV6diMYAf
> CSeq: 3136 REGISTER
> User-Agent: PJSUA v2.4.5 Linux-3.13.0.45/x86_64/glibc-2.19
> Supported: outbound, path
> Contact: <sip:911126592065 at 192.168.1.186:41924;transport=TLS;ob>;reg-id=1;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0000aa013800>"
> Expires: 300
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
> NOTIFY, REFER, MESSAGE, OPTIONS
> Authorization: Digest username="911126592065", realm="cdot.com",
> nonce="9064f7d7599a0682a98a0d3a2c6f78f3",
> uri="sips:192.168.103.175;transport=tls",
> response="9e6b6adaf30cd737396fbfcb852aa6c0", opaque="DEADBEEF"
> Content-Length:  0
>
>
> --end msg--
> 15:05:21.581   pjsua_core.c  .RX 498 bytes Response msg
> 200/REGISTER/cseq=3136 (rdata0x1746568) from TLS 192.168.103.175:5061:
> SIP/2.0 200 OK
> Call-ID: BZqPf3rFRXFueLb.LcxrL0hWV6diMYAf
> CSeq: 3136 REGISTER
> From: <sip:911126592065@192.168.103.175>;tag=9R61SCQehk-ZLda4TP-QVboszhEcyqU0
> To: <sip:911126592065 at 192.168.103.175>;tag=SBC+1+16f0001+11f7a1f9
> Via: SIP/2.0/TLS
> 192.168.1.186:41924;received=192.168.1.186;rport;alias;branch=z9hG4bKPjDhTOMSWiHtXVLW5tKMonLGlaSVtHUoEX
> Server: SIP/2.0
>
>
>
>
> But on sending INVITE pjsip is sending it on UDP instead of TLS as follows:
>
> INVITE sip:26592062 at 196.1.106.100 SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.1.186:5060;rport;branch=z9hG4bKPjOZ.C82g3um5IKRga.0iHurHd6RHgCjjr
> Max-Forwards: 70
> From: sip:911126592065@192.168.103.175;tag=6CSezVbYeuOivTtDHPfIJc2TCgXDq163
> To: sip:26592062 at 196.1.106.100
> Contact: <sip:911126592065 at 192.168.1.186:41924;transport=TLS;ob>
> Call-ID: p6u6bXAUKmLtakeDYszBE8vqht8Cz5xO
> CSeq: 16064 INVITE
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE,
> NOTIFY, REFER, MESSAGE, OPTIONS
> Supported: replaces, 100rel, timer, norefersub
> Session-Expires: 1800
> Min-SE: 90
> User-Agent: PJSUA v2.4.5 Linux-3.13.0.45/x86_64/glibc-2.19
> Content-Type: application/sdp
> Content-Length:   476
>
> v=0
> o=- 3658124143 3658124143 IN IP4 192.168.1.186
> s=pjmedia
> b=AS:84
> t=0 0
> a=X-nat:0
> m=audio 4000 RTP/AVP 98 97 99 104 3 0 8 9 96
> c=IN IP4 192.168.1.186
> b=TIAS:64000
> a=rtcp:4001 IN IP4 192.168.1.186
> a=sendrecv
> a=rtpmap:98 speex/16000
> a=rtpmap:97 speex/8000
> a=rtpmap:99 speex/32000
> a=rtpmap:104 iLBC/8000
> a=fmtp:104 mode=30
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:96 telephone-event/8000
> a=fmtp:96 0-16
>
>
> Please help
>
> regards
>
> shubham
>
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