[SIPForum-discussion] issue of the VoIP for the point-of-sale terminals

Wed Jan 23 03:39:12 UTC 2013

Hi All,

Thank you for your comments.

If it's defined in the RFC3264, I have the same questions as Murat, like 
the codec selection, and especially the binding between FXS port and RTP 
port. If the binding is done at the very beginning, we didn't the callee's 
RTP port, and so we don't know where to send the RTP packets. it will 
cause that part of our RTP stream are dropped. The handshake for the RTP 
layer will still fail if there is no repeatable mechanism.

That's really unreasonble.

BRs,

Joey

murat buker <murat_buker at yahoo.com> 
01/23/2013 08:32 AM
Please respond to
murat buker <murat_buker at yahoo.com>

To
Stephen James <sjames_1958 at yahoo.com>, Joey ZHENG 
<joey.zheng at sagemcom.com>
cc
"discussion at sipforum.org" <discussion at sipforum.org>
Subject
Re: [SIPForum-discussion] issue of the VoIP for the point-of-sale 
terminals

Thanks. Still I don't get it. If offerer offers 3 different codecs ( 
example g729,g711,g722), how can it allocate its dsp resource... As if it 
is g729 or as if it is g711 or as if it is g722 or for all of them 
separately ? It is not a resource wise behavior and not logical in my 
opinion.

But again thanks, I will research little bit more.

Murat Buker

From: Stephen James <sjames_1958 at yahoo.com>
To: murat buker <murat_buker at yahoo.com>; Joey ZHENG 
<joey.zheng at sagemcom.com> 
Cc: "discussion at sipforum.org" <discussion at sipforum.org> 
Sent: Wednesday, January 23, 2013 2:01 AM
Subject: Re: [SIPForum-discussion] issue of the VoIP for the point-of-sale 
terminals

My mistake it is RFC 3264 - 

Once the offerer has sent the offer, it MUST be prepared to receive
   media for any recvonly streams described by that offer.  It MUST be
   prepared to send and receive media for any sendrecv streams in the
   offer, and send media for any sendonly streams in the offer

Stephen James 
sjames_1958 at yahoo.com

We are not princes of the earth, we are the descendants of worms, and any 
nobility must be earned.

From: murat buker <murat_buker at yahoo.com>
To: Stephen James <sjames_1958 at yahoo.com>; Joey ZHENG 
<joey.zheng at sagemcom.com>
Cc: "discussion at sipforum.org" <discussion at sipforum.org>
Sent: Tue, January 22, 2013 5:47:06 PM
Subject: Re: [SIPForum-discussion] issue of the VoIP for the point-of-sale 
terminals

Hi James,

Early media means early SDP handsakes. In Joey's case, there is no SDP 
handshake, so gateway can't prepare itself ( because gateway does'nt know 
codec etc.., it can't allocate it's dsp resource). Please highlight RFC 
statements that cover your opinion, and let me correct myself.

Murat Buker

From: Stephen James <sjames_1958 at yahoo.com>
To: Joey ZHENG <joey.zheng at sagemcom.com> 
Cc: "discussion at sipforum.org" <discussion at sipforum.org> 
Sent: Tuesday, January 22, 2013 10:20 PM
Subject: Re: [SIPForum-discussion] issue of the VoIP for the point-of-sale 
terminals

Once the gateway has sent it should be prepared to receive RTP per RFC 
3261 or 3262. There may be a gateway setting to enable early media. 

Sent from my iPhone

On Jan 21, 2013, at 21:48, Joey ZHENG <joey.zheng at sagemcom.com> wrote:

Hi All, 

Now we got a problem: 
The test setup: 
POS payment terminal----->Gateway<---------->Internet<---->party B 
Once B received the invite, it will send the RTP packets immediately to 
Gateway. 
And it cause the problem that: 
Gateway didn't receive the SDP in 200 or 180 to know the RTP information, 
then early RTP packets to the gateway are dropped, not transferred to the 
POS terminal. 
But there are some handshake signal in these early RTP packets, like 
2100Hz CED. And since POS didn't receive the CED signal, the handshake 
between POS terminal and the party B always failed. 

Is there anyone who got the same problem, and what's the best solution for 
this? Is there any way to notify the party B to delay the RTP stream by 
the SIP message? 

BRS, 

Joey 

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