Linda Xiao wrote:
Hello,
See my reply inline....
P.s. is there anyone get some ideals on CODEC negotiation between SER and
client?
SER does not negotiate CODECs. UAs do. It does not do any transcoding
either as it is not a B2BUA.
Regards/Linda
-----Original Message-----
From: Siddhartha Roychowdhury [mailto:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> ]
Sent: Wednesday, September 22, 2004 1:01 PM
To: 'Linda Xiao'
Subject: RE: [Sip-implementors] NAT traversal for RTP stream
Hi,
The port numbers and IP addresses are correct. There seems to be a problem
with audio codec negotiation.
te2sp1 sends an INVITE saying that it can support 4, 0 or 8
SER replies in 200 OK that it can support 0, 8 or 18
[Linda] 18 should not be in 200ok.
ak2sp1 sends an INVITE with codec types as 0, 18, 101, 102 ...... SER
replies in 200 OK that it can support 0, 8, 18 or 103
ak2sp1 transmits G.729 packets (codec 18), te2sp1 transmits G.711 (codec 0).
I'm not very familiar with the whole codec transcoding thing but I thing
this is what is causing the one way audio problem.
[Linda] Do you mean that CODEC 18 cause the problem??? I cannot see the
reason behind. I will see if I can force "ak2sp1" transmitting packets in
CODEC 0 format.
I think the 200 OK should have carried only 0 since that's the only codec
that's common between what the 2 UA's offered,
[Linda] The 2 UA's are independent units. It is not necessary to restrict to
return the common supported CODECs from SER. Thus, the returning for
"ak2sp1" is correct, but I do not understand why type 18 for "te2sp1".
unless SER does the transcoding properly, which it's not. That's my theory.
Siddhartha
-----Original Message-----
From: Linda Xiao [mailto:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> ]
Sent: Wednesday, September 22, 2004 2:28 PM
To: 'Siddhartha Roychowdhury'
Cc: '[EMAIL PROTECTED]'; '[EMAIL PROTECTED]'
Subject: RE: [Sip-implementors] NAT traversal for RTP stream
Hello Siddhartha,
Thank you for your reply.
Yes. The SIP proxy is SER, and it is located on the internet which is out of
my control. Unfortunately, I cannot trace from the proxy side.
From the ethereal dump, I guest the communication between SIP proxy & RTP
proxy is functional properly, otherwise, the "ak2sp1" would receive a sound
correct modified SDP in 200OK. I doubt it is the SER issue, otherwise,
"te2sp1" cannot be OK. I suspect that there must be something wrong with the
info "ak2sp1" sent out that confused the RTP proxy, most likely, the RTP
proxy was listening to a wrong IP:port.
Regards/Linda
-----Original Message-----
From: Siddhartha Roychowdhury [mailto:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> ]
Sent: Wednesday, September 22, 2004 11:50 AM
To: 'Linda Xiao'
Subject: RE: [Sip-implementors] NAT traversal for RTP stream
Hi,
It's not obvious from the traces what the problem is. The traces are for
packets transmitted on the UA/NAT device network. I think you are using Sip
Express Router(SER) as the proxy. I think u need to sniff on the network
between the NAT device and the proxy to see where the RTP packets from the
UA are getting dropped - at the proxy or even before that. SER has a module
called nathelper that needs to be installed for NAT traversal. I'm assuming
that your installation already has that - otherwise you would not have 2 way
audio for one UA. If the packets are not reaching the proxy, it's a network
routing/configuration problem. If it's an SER problem, you can post this
problem on their website.
Siddhartha
-----Original Message-----
From: Linda Xiao [mailto:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> ]
Sent: Wednesday, September 22, 2004 11:21 AM
To: 'Siddhartha Roychowdhury'
Subject: RE: [Sip-implementors] NAT traversal for RTP stream
There might be something wrong with the mail server!!??
Here you go.
-----Original Message-----
From: Siddhartha Roychowdhury [mailto:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> ]
Sent: Wednesday, September 22, 2004 9:20 AM
To: 'Linda Xiao'
Subject: RE: [Sip-implementors] NAT traversal for RTP stream
Hi! I think attachments sent on the mailing list are stripped off. Could you
send me a the zip-ed file(s)? I worked on the NAT traversal solution for my
company. I would like to take a look.
Thanks,
Siddhartha
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> ] On Behalf Of Linda Xiao
Sent: Wednesday, September 22, 2004 11:04 AM
To: '[EMAIL PROTECTED]'; '[EMAIL PROTECTED]'
Subject: RE: [Sip-implementors] NAT traversal for RTP stream
Strange!? Anyway, here is the attachment.
Thanks & regards/Linda
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> ]
Sent: Tuesday, September 21, 2004 8:45 PM
To: [EMAIL PROTECTED]
Subject: RE: [Sip-implementors] NAT traversal for RTP stream
There are no attachments send..
Can u send them again.
-sreeram
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> ] On Behalf Of Linda Xiao
Sent: Wednesday, September 22, 2004 12:28 AM
To: '[EMAIL PROTECTED]'; '[EMAIL PROTECTED]'
Subject: [Sip-implementors] NAT traversal for RTP stream
Hi all,
I have 2 Uas setup behind a NAT. After a call is initiated, "te2sp1" can
build 2-way rtp traffic while "ak2sp1" cannot. Attached please find the
ethereal dump. By comparing of these 2 dumps, I cannot find out the reason.
Can anyone identify why server side stopped RTP transmission to "ak2sp1"?
Any comment is appreciated.
<<ak2sp1.zip>> <<te2sp1.zip>>
Regards/Linda
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