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Today's Topics:
1. Re: Is "fmtp" line needed in 200 OK answer?
(Andrea Puddu)
2. Tutorials on SIP (and other protocols) over at
Techtionary.com (Dan York)
3. Re: Prefered audio codec (Jeff Wright)
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Message: 1
Date: Wed, 12 Dec 2007 10:38:15 +0000
From: Andrea Puddu <androjokerhotmail.com>
Subject: Re: [SIPForum-discussion] Is "fmtp" line
needed in 200 OK
answer?
To: Herve Jourdain <herve.jourdainmstarsemi.com>,
"'Donald Lee'"
<baolovebaogmail.com>
Cc: discussionsipforum.org
Message-ID: <BLU124-W30CB97029A614DEC2C41BFC0650phx.gbl>
Content-Type: text/plain; charset="windows-1252"
Thanks for your analysis.
By chance, I've just found a parameter on the phone that can
help me -> "Full SDP Answer" .... I've
switched it to OFF.
Now the phone sends only one codec in the 200 OK response
(g729), but without the "fmtp" line.
So the phone starts talking in G729 but AS (or SBC) sends a
BYE to the phone!!!!!
I'm starting to think that SBC does not like the lack of
"fmtp" filed in the response .....
Andrea
From: herve.jourdainmstarsemi.com
To: androjokerhotmail.com; baolovebaogmail.com
CC: discussionsipforum.org
Subject: RE: [SIPForum-discussion] Is "fmtp" line
needed in 200 OK answer?
Date: Wed, 12 Dec 2007 11:25:55 +0100
Hi,
As far as I remember of
RFC 3264, the devices should be prepared to receive ANY of
the codecs it sends
in its offer/response?
So basically, if you
respond with 3 codecs, you should be able to receive at any
time on any of
those 3 codecs, and switch between them?
That?s why it seems
customary on several phones to answer with only one codec?
But even if it?s
stated this way in the specs, I still do think that before
switching to another
codec, the UAs should at least use the codec that was agreed
for in the negotiation
at first (the ?highest priority? one, in your case G729).
Experience shows,
unfortunately, that even if it?s usually the case, it?s NOT
ALWAYS
the case?
I think you might have an
example here, and I know I?ve met several devices ? phones
or
gateways usually ? that would also behave in this way?
So maybe try with only 1
codec in the response, to see if it works better?
But if you do so, please
switch to outband DTMF (RFC 2833 or RFC 4733), because once
you give only G729
in the answer there should not be any possibility to switch
to G711 for in-band
DTMF.
And your SDP suggests
outband DTMF is not activated.
Regards,
Herve
From:
discussion-bouncessipforum.org
[mailto:discussion-bouncessipforum.org] On Behalf Of
Andrea Puddu
Sent: mercredi 12 d?cembre 2007
09:35
To: Donald Lee
Cc: discussionsipforum.org
Subject: Re: [SIPForum-discussion]
Is "fmtp" line needed in 200 OK answer?
I agree with your analysis. I've noticed that for example
Cisco and Linksys phone sends only one codec in the 200 OK
SDP envelope.
But... Can the sending of multiple codec in the 200 OK
response be an issue?
I cannot complain with phone provider about this behaviour.
Anyway why does the phone talk with G729 and AS talk with
G711? I'm
wondering if the AS performs a check on fmtp field in the
200 OK response
.................
Thanks,
Andrea
Date: Wed, 12 Dec 2007 16:24:05
+0800
From: baolovebaogmail.com
To: androjokerhotmail.com
Subject: Re: [SIPForum-discussion] Is "fmtp" line
needed in 200 OK
answer?
CC: discussionsipforum.org
# the "g" and "G" have no difference.
and the offer indicate that it's not G729B codec.
# I don't think the lack of fmtp in answer is the reason
for issue. Maybe the ip phone should reply only one codec in
its SDP. Maybe the
multi codec in answer make the offer confused.
On 12/11/07, Andrea Puddu <androjokerhotmail.com> wrote:
Hello guys,
I'm facing an issue when a mobile (common mobile 3G phone)
tries to call an
internal SIP phone through its geographical number.
The issue is that when the SBC sends the INVITE to the IP
phone, in the
SDP enclosure it suggests these codecs:
m=audio 19570 RTP/AVP 18 8 0
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
The 200 OK of the phone (SDP part only) is:
m=audio 57956 RTP/AVP 18 8 0
a=rtpmap:18 g729/8000
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=sendrecv
So the IP phone start to talk with G729 and SBC replies with
G711A!!! So I
can't hear the voice coming from mobile phone!!!!
I have two questions:
- Can the difference between the offer (G729) and the answer
(g729) be
significant? I mean the difference because the "g"
is not capital in
the answer
- Can the lack of the "fmtp" line in the answer
cause troubles?
Thanks 1000,
Andrea
Prenditi una pausa e sfida i tuoi amici a Ladybird su
Messenger! Messenger Giochi
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--
BR
Donald
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Live Messenger
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Message: 2
Date: Wed, 12 Dec 2007 08:05:18 -0500
From: Dan York <dyorkvoxeo.com>
Subject: [SIPForum-discussion] Tutorials on SIP (and other
protocols)
over at Techtionary.com
To: discussionsipforum.org
Message-ID: <475FDC8E.4010303voxeo.com>
Content-Type: text/plain; charset=ISO-8859-1
As there have been a lot of questions on this list lately
asking about
basic aspects of SIP, I thought I'd just mention a resource
that's out
there. Tom Cross has assembled a whole bunch of Flash-based
tutorials
at http://www.techtionary.co
m/ on a wide range of tutorials.
Unfortunately, since his site is based on Flash, I can't
give you a URL
to go directly to the SIP-related tutorials, but if you go
to
techtionary.com and click on "S", you can then
scroll down to the
various SIP-related tutorials. In many of them he has
provided
graphical illustrations of the call flow for various aspects
of SIP. He
also has tutorials on SRTP and a wide range of other
non-VoIP-related
protocols.
Regards,
Dan
P.S. I have no affiliation with Techtionary.com and cannot
vouch for the
accuracy or anything else about the site. I just know of
Tom and the
site and thought it might be of use for some folks looking
to learn
about SIP.
--
Dan York, CISSP, Director of Emerging Communication
Technology
Office of the CTO Voxeo Corporation dyorkvoxeo.com
Phone: +1-407-455-5859 Skype: danyork http://www.voxeo.com
Blogs: http://blogs.voxeo.com http://www.disrupt
ivetelephony.com
Bring your web applications to the phone.
Find out how at http://evolution.voxeo.com
------------------------------
Message: 3
Date: Wed, 12 Dec 2007 11:19:35 -0500
From: "Jeff Wright" <JWrightazteknetworks.net>
Subject: Re: [SIPForum-discussion] Prefered audio codec
To: "Fredrik Sandedal" <fredrik.sandedalenera.se>,
<discussionsipforum.org>
Message-ID:
<ABFF960C9FAE494482BD527AB5D6263F0820EF8CMI8NYCMAIL03.Mi8.com>
Content-Type: text/plain; charset="us-ascii"
My experience in the past has shown that G.729a will do a
better job
than most LBRCs (low bit rate codecs) of passing DTMF digits
(and tones
in general). It certainly is a lot better than G.723.1.
But don't bet
the bank on it. Make sure you have "wide" digits
(i.e. at least 100ms
duration both on and off) from the phone device.
Is there any reason you have to pass the digits through the
voice path?
Why not use some digit relay protocol like RFC4733?
Jeffrey D. Wright
System Test Engineering Manager
Aztek Networks, Inc.
(303) 415 6149
jwrightazteknetworks.net
________________________________
From: discussion-bouncessipforum.org
[mailto:discussion-bouncessipforum.org] On Behalf Of
Fredrik Sandedal
Sent: Tuesday, December 11, 2007 12:50 AM
To: discussionsipforum.org
Subject: [SIPForum-discussion] Prefered audio codec
Hi,
Could someone give me suggestions on the best audio codec to
use when
DTMF is necessary and the bandwidth must be kept on a low
rate?
I heard that there could be problems with DTMF and
G729AnnexA but I
haven't been able to do any advanced test with this.
Regards,
Fredrik Sandedal, Enera
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End of discussion Digest, Vol 29, Issue 29
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