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  1. #1
    Registered User
    Join Date
    Nov 2007

    SIP Codec Error (-17767)

    I writhed a small SIP Client without profile by nokia , but when i select register function i have this error:
    KErrSipCodecPort -17767

    In this forum i don't find any post with SIP Codec error ...

    my Register header fields is created with this functions
    CSIPToHeader* aor = CreateToHeaderL(_L8("sip:349@serveraddres"));
    CSIPContactHeader* contact = CreateContactHeaderL(_L8("sip:349@localaddress:5060"));
    CUri8* remoteUri = ConvertToUri8LC(_L8("sip:serveraddress:5060"));
    CSIPFromHeader* from = CreateFromHeaderL(_L8("sip:349@serveraddress"));

    my regostration binding is created succesufuly but when i call RegisterL() i take error

    CSIPClientTransaction* iClientTransaction = iRegistrationBinding->RegisterL(); // System Error (-17767)

    in Documentation i don't find any kind of Sip Codec error, and i don't find how create "Via: " header field, if is create automaticaly how change parameters ? .

    thanks !

  2. #2
    Regular Contributor
    Join Date
    Aug 2007

    Re: SIP Codec Error (-17767)


    It is not possible to create the Via-header, it is automatically generated by the SIP stack.

    The -17767 error occurs when decoding an URI, and the port is corrupted. Does the creating of aor, remote uri, contact & from go successfully? What kind of localaddress and serveraddress values are you passing to them?


  3. #3
    Registered User
    Join Date
    Nov 2007

    Re: SIP Codec Error (-17767)

    Tnx , problem was in uri

    Now i have other problem , when i send INVITE , server send 100 Tryng , then 403 Forbbiden , i don't know what is wrong in my paket.

    i make call from 349 to 339 , and server address is: , and local address:

    Session Initiation Protocol
    Request-Line: INVITE sip:339@ SIP/2.0
    Method: INVITE
    [Resent Packet: False]
    Message Header
    Via: SIP/2.0/UDP;branch=z9hG4bK0iui25b7v1hc795nbidjhu4
    Transport: UDP
    Sent-by Address:
    Sent-by port: 5060
    Branch: z9hG4bK0iui25b7v1hc795nbidjhu4
    From: 349 <sip:349@>;tag=p6l225drcdhc7bncbidq
    SIP Display info: 349
    SIP from address: sip:349@
    SIP tag: p6l225drcdhc7bncbidq
    To: <sip:339@>
    SIP to address: sip:339@
    Contact: <sip:349@>
    Contact Binding: <sip:349@>
    URI: <sip:349@>
    SIP contact address: sip:349@
    CSeq: 693 INVITE
    Sequence Number: 693
    Method: INVITE
    Call-ID: b34x5teeoIcvsROrO44amt-iITf5ZE
    Supported: sec-agree
    Max-Forwards: 70
    Content-Type: application/sdp
    Content-Length: 148
    Message Body
    Session Description Protocol
    Session Description Protocol Version (v): 0
    Owner/Creator, Session Id (o): TopexPhone 196678571 0 IN IP4
    Session Name (s): Topex SIP Call
    Connection Information (c): IN IP4
    Connection Network Type: IN
    Connection Address Type: IP4
    Connection Address:
    Time Description, active time (t): 0 0
    Session Start Time: 0
    Session Stop Time: 0
    Media Description, name and address (m): audio 0 RTP/AVP 8
    Media Type: audio
    Media Port: 0
    Media Proto: RTP/AVP
    Media Format: ITU-T G.711 PCMA
    Media Attribute (a): rtpmap:8 PCMA/8000
    Media Attribute Fieldname: rtpmap
    Media Format: 8
    MIME Type: PCMA

    i compare my pakets with SJPhone client and i find 2 differences :
    1) Call_ID is different , but i don't know how change it and i don't know how attach host.
    2) CSeq in my INVITE packet continue sequense from Register packets, but in RFC he must start from 1 , i there a importance ?

    Thanks !

  4. #4
    Registered User
    Join Date
    Nov 2007

    Re: SIP Codec Error (-17767)

    problem is resolved ..

  5. #5
    Registered User
    Join Date
    Jun 2008

    Re: SIP Codec Error (-17767)

    Quote Originally Posted by Nicolae View Post
    problem is resolved ..
    How did you resolved the problem with 403 Forbidden?I have the same problem with my SIP Client for Nokia!

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