When a SIP endpoint sends its codecs, it includes the ptime. But, only
the first ptime on the list is contained in the SDP. This just caused a
problem for me because of the following:
My Snom phone only allows 1 sample size setting per codec, and it's
configured for 20ms. The endpoint calling the snom is configured for 10
and 20ms, where 10 will be listed first in the SDP.
As you can see below, the ptime is 10ms. But, the second rtpmap is for
G729A 20ms. This is where the current RFC, 4566 is faulty. It should
contain a list of all applicable Ptime variations and not just the first
one. Otherwise, this scenario will always fail.
Does anyone agree? If so, how do I go about making a recommendation to
modify the RFC?
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): - 1171987362 1171987362 IN
IP4 10.153.96.5
Session Name (s): Aspect Communications SDP Session
Connection Information (c): IN IP4 10.153.96.5
Time Description, active time (t): 0 0
Media Description, name and address (m): audio 10034 RTP/AVP
18 18 101
Media Title (i): telephone-event 8000
Connection Information (c): IN IP4 10.153.96.5
Media Attribute (a): sendrecv
Media Attribute (a): ptime:10
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute (a): fmtp:18 annexb=no
Media Attribute (a): rtpmap:18 G729/8000
Media Attribute (a): fmtp:18 annexb=no
Media Attribute (a): rtpmap:101 telephone-event/8000
Thank you,
Steve Vick
QA Advisor, Senior
Aspect Software
829 Parkview Boulevard
Lombard, Illinois 60148
o (630) 227-7426
c (630) 699-7257
mailto:[EMAIL PROTECTED]
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