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

Wed Jan 23 13:18:57 UTC 2013

The local RTP port is known as that is in the SDP offered. 

Also, reading RFC 3960

UACs are ready to play incoming media packets as soon as    they send an offer, 
because they cannot count on the reception of the    200 (OK) to start playing 
out media for the caller; SIP signalling    and media packets typically traverse 
different paths, and so, media    packets may arrive before the 200 (OK) 
response.

Previously our system did not allow media prior to receipt of the far end SDP. 
Per a customer request we implemented the ability receive RTP that matched our 
offer prior to receipt of the answer to 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: Joey ZHENG <joey.zheng at sagemcom.com>
To: murat buker <murat_buker at yahoo.com>; "discussion at sipforum.org" 
<discussion at sipforum.org>; Stephen James <sjames_1958 at yahoo.com>
Sent: Tue, January 22, 2013 9:39:27 PM
Subject: Re: [SIPForum-discussion] issue of the VoIP for the point-of-sale 
terminals

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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