RFC3551 (RTP Profile)
-----------------------
   This profile reserves payload type numbers in the range 96-127
   exclusively for dynamic assignment.  Applications SHOULD first use
   values in this range for dynamic payload types.  Those applications
   which need to define more than 32 dynamic payload types MAY bind
   codes below 96, in which case it is RECOMMENDED that unassigned
   payload type numbers be used first.  However, the statically assigned
   payload types are default bindings and MAY be dynamically bound to
   new encodings if needed.  Redefining payload types below 96 may cause
   incorrect operation if an attempt is made to join a session without
   obtaining session description information that defines the dynamic
   payload types.-----------------------------------------------

According to RFC dynamic payload type can be assigned even with number below 
96, however I have never seen such an implementation which uses dynamic payload 
type < 96, it will bring a lot of incompatibility issues. So I believe the 
following routine is robust enough to match the codec.

if payload type <  96, match payload type number only
otherwise if payload type is in range [96-127] , match by codec name & sampling 
rate & channel count

This will help to avoid mess concerning codec names like PCMU <-> g711u; g729 
<-> g729a (from RTP perspective it's just the same codec) at least in case of 
static mapping.

Hope this will help.
Thanks,
Arsen.

----- Original Message ----
From: Jerry Richards <[EMAIL PROTECTED]>
To: sofia-sip-devel@lists.sourceforge.net
Sent: Tuesday, July 10, 2007 1:44:18 AM
Subject: Re: [Sofia-sip-devel] Incoming Re-INVITE SDP Content

Do you have an RFC reference that states that the name shouldn't be compared
if the number is 96 or less?  I don't know enough about SDP to confirm that.

Thanks,
Jerry


-----Original Message-----
From: Michael Jerris [mailto:[EMAIL PROTECTED]
Sent: Monday, July 09, 2007 12:32 PM
To: Pekka Pessi; Jerry Richards
Cc: sofia-sip-devel@lists.sourceforge.net
Subject: Re: [Sofia-sip-devel] Incoming Re-INVITE SDP Content


Shouldn't it be using just the number for 96 and less, and only compare the
name if >= 97?

Mike


On 7/9/07 8:08 PM, "Pekka Pessi" <[EMAIL PROTECTED]> wrote:

> 2007/7/6, Jerry Richards <[EMAIL PROTECTED]>:
>> In response to a re-INVITE from the server, I am seeing sofia-sip (v
1.12.4)
>> sending out a 200 OK SDP body that appears to be missing a field.  Here
are
>> the details:
>
>> 1) NUA sends outgoing INVITE (via nua_invite() with SOATAG_USER_SDP)
>>    containing the following (as seen in Ethereal):
>>                 ... (m): audio ...
>>                 ... (a): rtpmap:0 G711/8000
>
>
>> 2) Server replies (via 200 OK) with SDP fields (as seen in Ethereal):
>>                 ... (m): audio ...
>>                 ... (a): rtpmap:0 PCMU/8000
>>                 ... (a): rtpmap:96 telephone-event/8000
>>                 ... (a): fmtp:96 0-16
>>                 ... (a): silenceSupp:off - - - -
>>
>> 3) Server sends re-INVITE with SDP fields (as seen in Ethereal):
>>                 ... (m): audio ...
>>                 ... (a): rtpmap:0 PCMU/8000
>>                 ... (a): rtpmap:96 telephone-event/8000
>>                 ... (a): fmtp:96 0-16
>>                 ... (a): silenceSupp:off - - - -
>>
>> 4) NUA replies (via nua_respond(200 "OK") and identical SOATAG_USER_SDP
>>    as original INVITE), but this is what I see in outgoing 200 OK SDP
>>    body in Ethereal:
>>                 ... (m): audio ...
>>                 ... (a): rtpmap:96 telephone-event/8000/1
>>
>> The outgoing 200 OK should also contain either
>>
>>                 ... (a): rtpmap:0 PCMU/8000
>>                                 or
>>                 ... (a): rtpmap:0 G711/8000
>>
>> I have a Polycomm phone that I am comparing this with.  Do you know why
this
>> field is missing?
>
> I guess the "G711" codec name is your problem. If they payload type
> name is there, the soa and sdp uses the name and not the payload type
> number when it matches the codecs. So "G711" does not match with
> "PCMU".
>
> Polycomm either digs "G711" in offer but uses "PCMU" in answer, or
> rejects all your codecs but includes PCMU in answer in distant hope
> that your UA could send PCMU to it.
>
> SOA doesn't include G711, probably because there is the
> telephony-event codec which it selects as common codec. I guess if you
> included a tag SOATAG_AUDIO_AUX("telephone-event") in
> nua_set_params()/nua_invite() tags, SOA would include both "G729" and
> "G711" in the answer.


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