[SIPForum-discussion] Need Detail on SIP to RTP conversion Point

[rajesh]

Hi 
Thanks for the response.
u mean to say that based upon the via field 400 response will travel back to UAC from proxy.


Thanks and Regards
Rajesh Kumar
Sr. Software Engineer
R & D - Network Group 
+91 40 23555945 - 235
+91 99084 00027
www.imimobile.com 


  ----- Original Message ----- 
  From: Sharanagoud BD. 
  To: Halit Sakca ; rajesh ; discussion 
  Sent: Thursday, April 10, 2008 10:09 AM
  Subject: RE: [SIPForum-discussion] Need Detail on SIP to RTP conversion Point


  Hi Rajesh,

  If there is no ip address in from field also, error response will reach the client.
  bcz always responses depends on the "Via" field. So 400 error response will be the
  answer.

  Thanks,
  Sharanagoud.
     






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  From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of Halit Sakca
  Sent: Wednesday, April 09, 2008 8:32 PM
  To: rajesh; discussion
  Subject: Re: [SIPForum-discussion] Need Detail on SIP to RTP conversion Point


  Hey Rajesh,
   
  "400 Missing Mandatory SIP Header" will be the response.

  Selamlar,
  Halit Sakca





----------------------------------------------------------------------------
    From: rajeshkumar.r at imimobile.com
    To: discussion at sipforum.org
    Date: Tue, 8 Apr 2008 16:31:17 +0530
    Subject: Re: [SIPForum-discussion] Need Detail on SIP to RTP conversion Point


    Hi All,
    what will be proxy response if we are sending an request with no parameter in From 
    field. and how the  error response will arrive back to client as there is  no IP address in from field.

    Thanks and Regards
    Rajesh Kumar
    Sr. Software Engineer
    R & D - Network Group 
    +91 40 23555945 - 235
    +91 99084 00027
    www.imimobile.com 


      ----- Original Message ----- 
      From: Rajeshkumar Babu 
      To: discussion 
      Sent: Sunday, April 06, 2008 2:50 PM
      Subject: [SIPForum-discussion] Need Detail on SIP to RTP conversion Point


      Hi

      Kindly tell me the point where the RTP headers get know the B Party IP address and start the RTP transactions for a voice call.

      Like in SS7 ISUP msg the CIC is taken from initial IAM message.

      So the B Party address is taken from contact of 200 ok messages or ....pls correct me...??

      -- 
      Thanks & Regards

      Rajeshkumar B


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