[SIPForum-discussion] Need Detail on SIP to RTP conversion Point
rajesh
rajeshkumar.r at imimobile.com
Thu Apr 10 04:57:17 UTC 2008
Hi ,
I have a doubt regarding branch and tag field.
tag is basically with to and from field.
branch is basically with via field.
Proxy recvs a request from UAC.
Proxy forks the request and add brach parameter to differentiate different response coming from UAS.
and same response gets transferred back to UAC.
In this complete scenario where the tag field will come into picture.
Please correct me if i am wrong.
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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