[SIPForum-discussion] Need Info - UAC behavior

[Karthik Ramiya]

Hi,

   I am looking for a softclient which supports all the variants of G729 namely G729, G729A, G729B and G729AB. If any of you guys came across such a client catering these requirements please suggest me.

TIA,
Karthik


________________________________
From: discussion-bounces at sipforum.org [discussion-bounces at sipforum.org] On Behalf Of RAJESH KUMAR RAJU [rajeshk.r at samsung.com]
Sent: Tuesday, January 27, 2009 10:35 AM
To: Moshe Ostrovsky
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Need Info - UAC behavior


Hi ,

Kindly find out ur sip stack follows which rfc . whether it is following rfc 4566 or RFC 3551 .



Thanks and Regards

Rajesh






------- Original Message -------
Sender : Moshe Ostrovsky<mosheo at radvision.com>
Date : Jan 25, 2009 05:02 (GMT+09:00)
Title : RE: [SIPForum-discussion] Need Info - UAC behavior

Hi All.

Regarding Payload type number, can you please point to the RFC part that sets the range validity restrictions?

What I could see in RFC 4566 is that there is no restriction.

According to RFC 3551 – the range is 0-127.

What happens if I need to define dynamic codec beyond 127?

Thanks.

From: discussion-bounces at sipforum.org [mailto:discussion-bounces at sipforum.org] On Behalf Of RAJESH KUMAR RAJU
Sent: Thursday, January 22, 2009 7:08 AM
To: Gowthaman Desingh
Cc: discussion at sipforum.org
Subject: Re: [SIPForum-discussion] Need Info - UAC behavior

Dear Gowtham,

*m=audio 6000 RTP/AVP 12345*

 *a=rtpmap:8 PCMA/8000*



Any Objective behind doing this , because as per rfc  , i have never seen any one doing like this . and 12345 must be a valid defined

Payliad type and also the Payload type number must be in the defined range . u can check the rfc .






------- Original Message -------
Sender : Gowthaman Desingh<gowtham_dgm at rediffmail.com>
Date : Jan 21, 2009 15:31 (GMT+09:00)
Title : [SIPForum-discussion] Need Info - UAC behavior

Hi All,
  I would like to know the expected behavior of the UAC in the below mentioned scenario.

Setup:
        |SIP Phone | -------- |sipp|
    I have connected a SIP phone back to back with SIPP.

1. I send an INVITE request from the SIP Phone to the sipp with the below mentioned SDP offer.
**********************************
v=0
o=SIPUA 22986 0 IN IP4 9.7.40.253
s=SIP Call
t=0 0
m=audio 19570 RTP/AVP 0 8 18 102 9 101
c=IN IP4 9.7.40.253
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
**********************************

2. Accept the call from sipp by sending 200 OK.
  In the 200 OK sent by the sipp, I send the below mentioned SDP answer. The payload number mentioned in the m=line is "12345"(which is invalid). But the payload number mentioned in the attribute is valid (8 - PCMA).
**********************************
v=0
o=user1 53655765 2353687637 IN IP4 9.7.40.50
s=-
t=0 0
a=recvonly
c=IN IP4 9.7.40.50
*m=audio 6000 RTP/AVP 12345*
c=IN IP4 9.7.40.50
*a=rtpmap:8 PCMA/8000*
a=sendonly
***********************************


My phone is sending 200 OK followed by BYE to terminate the call. Is that a valid behavior.

Thanks,
Gowthaman D

u can reach me on my mobile @ +919886625873
<http://adworks.rediff.com/cgi-bin/AdWorks/click.cgi/www.rediff.com/signature-home.htm/1050715198@Middle5/2703345_2674874/2678496/1?PARTNER=3&OAS_QUERY=null>


















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