hello,
I guess it is rejected by pjsip sdp negotiation code. Take a look
soure code , you will get the answer.
normally if we want to change rtp ip and port, it is enough to
replace ip and port at SDP and modify sdp version number.
It seems your remote peer re-INVITE SDP is copyed from other
device. I guess it want to disable old media stream and create new
one, but pjsip need media stream in original order.
regards,
Gang
On Wed, Nov 2, 2011 at 3:55 AM, Tom Johnson <TJohnson at microaut.com> wrote:
> I am getting a ?488 Not Acceptable Here? from my PJSIP application when the
> caller sends a second INVITE request. ??I have put breakpoints at the
> callbacks of my code and they do not get called when the request arrives.
> Below are the SIP messages:
>
>
>
> Initial Invite:
>
> INVITE sip:11015434001 at 172.25.53.92:5060 SIP/2.0
>
> To: <sip:11015434001 at 172.25.53.92>
>
> From: "Agent 9" <sip:1009@172.23.74.54>;tag=snl_L5KQCRbPEj
>
> Call-ID: SEC11-44a17ac-44b17ac-1-t4i4ZZEk3IGL
>
> CSeq: 1235 INVITE
>
> Contact: <sip:1009 at 172.23.74.54:5060;maddr=172.23.74.54>
>
> Via: SIP/2.0/UDP
> 172.23.74.54:5060;branch=z9hG4bKSEC-44a17ac-44b17ac-1-b0blO67CJ3
>
> Accept-Language: en;q=0.0
>
> Alert-Info:<Bellcore-dr1>
>
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
>
> P-Asserted-Identity: "Agent 9" <sip:1009 at 172.23.74.54>
>
> Supported: timer
>
> Date: Tue, 01 Nov 2011 19:45:48 GMT
>
> Max-Forwards: 69
>
> Content-Type: application/sdp
>
> Content-Length: 289
>
>
>
> v=0
>
> o=MxSIP 0 53993236 IN IP4 172.23.74.100
>
> s=SIP Call
>
> c=IN IP4 172.23.74.100
>
> t=0 0
>
> m=audio 5010 RTP/AVP 0 8 18 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=silenceSupp:off - - - -
>
> a=fmtp:18 annexb=no
>
> a=fmtp:101 0-15SIP/2.0 200 OK
>
> Via: SIP/2.0/UDP
> 172.23.74.54:5060;received=172.23.74.54;branch=z9hG4bKSEC-44a17ac-44b17ac-1-b0blO67CJ3
>
> Call-ID: SEC11-44a17ac-44b17ac-1-t4i4ZZEk3IGL
>
> From: "Agent 9" <sip:1009@172.23.74.54>;tag=snl_L5KQCRbPEj
>
> To: <sip:11015434001 at 172.25.53.92>;tag=Lq8Ky21ENaWJaCCdAtdNAa1Fd.kK8Ta9
>
> CSeq: 1235 INVITE
>
> Contact: <sip:172.25.53.92:5060>
>
> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE
>
> Supported: replaces, 100rel, timer
>
> Content-Type: application/sdp
>
> Content-Length:?? 210
>
>
>
> v=0
>
> o=pjsip-siprtp 3529165547 3529165548 IN IP4 mbpdeb
>
> s=pjsip
>
> c=IN IP4 172.25.53.92
>
> t=0 0
>
> m=audio 4000 RTP/AVP 0 121
>
> a=rtpmap:0 PCMU/8000
>
> a=sendrecv
>
> a=rtpmap:121 telephone-event/8000
>
> a=fmtp:121 0-15
>
>
>
>
>
> Response:
>
>
>
>
>
> Re-Invite:
>
> INVITE sip:172.25.53.92:5060 SIP/2.0
>
> To: <sip:11015434001 at 172.25.53.92>;tag=Lq8Ky21ENaWJaCCdAtdNAa1Fd.kK8Ta9
>
> From: "Agent 9" <sip:1009@172.23.74.54>;tag=snl_L5KQCRbPEj
>
> Call-ID: SEC11-44a17ac-44b17ac-1-t4i4ZZEk3IGL
>
> CSeq: 1236 INVITE
>
> Contact: <sip:189*12147 at 172.23.74.54:5060;maddr=172.23.74.54>;isfocus
>
> Via: SIP/2.0/UDP
> 172.23.74.54:5060;branch=z9hG4bKSEC-44a17ac-44b17ac-1-J0bMn816tg
>
> Accept-Language: en;q=0.0
>
> Allow: REGISTER, INVITE, ACK, BYE, CANCEL, NOTIFY, REFER, INFO
>
> P-Asserted-Identity: "Agent 9" <sip:1009 at 172.23.74.54>
>
> Supported: timer
>
> Date: Tue, 01 Nov 2011 19:45:54 GMT
>
> Max-Forwards: 69
>
> Content-Type: application/sdp
>
> Content-Length: 374
>
> Allow-Events: conference
>
>
>
> v=0
>
> o=- 2271596941 2271596941 IN IP4 172.23.74.71
>
> s=MS-Ver.33.5.10.1-990
>
> c=IN IP4 172.23.74.71
>
> t=0 0
>
> m=audio 20016 RTP/AVP 0 8 18 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:101 telephone-event/8000
>
> m=audio 0 RTP/AVP 0 8 18 101
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:8 PCMA/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:101 telephone-event/8000
>
>
>
> Am I missing something on the initialization??? I do not see how it can be
> anything else since my code is not being called.? Note: I originally tried
> this on v1.10, but these are from v2.0-alpha2.
>
>
>
> Thomas G. Johnson
>
>
>
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