[asterisk-dev] Packetization

Dan Austin Dan_Austin at Phoenix.com
Tue Feb 28 11:04:02 MST 2006


I have an issue about packetization/frame length customization for the
RTP stream generation. 
 
 I needed to tune up the hardcoded packetization value of Asterisk
(which is 20ms). Consequently I applied the patch located at
http://bugs.digium.com/view.php?id=5162. 
 
 Patch worked and I am now able to customize packetization value per
user/peer basis. However,

	*	packetization parameter setting has no effect if used at
the [default] part of sip.conf, am I correct?   I performed several
tests with SIP clients from Grandstream. Please note the below; 
	*	models are HT & GXP with the latest firmware releases 
	*	Asterisk version is 1.2.4 
	*	choice of codec is G.729 
	*	configured frame length is 50ms at the clients and
Asterisk 
	*	clients register to Asterisk with reinvite=no option for
RTP proxying.

	Grandstream clients fail to send RTP stream with 50ms customized
frame rate. Instead, they send with the default frame length which is
20ms. However, Asterisk sends with 50ms, as configured. As a result, RTP
flows with asymmetric bandwidth values, that is; 37Kbps from the client
and 20.5Kbps from the Asterisk.

	I trunked two Asterisk systems using SIP and noticed that RTP
flows perfectly, with 50ms/50ms frame lengths to/from any Asterisk.

	*	When I checked the signaling with the packet sniffer, I
noticed that Grandstream implements ptime attribute in SDP and
explicitly declares frame length in the INVITE msg. However, 200 OK msg
from Asterisk does not include the confirmation of the ptime value,
resulting in Grandstream's decision to default to 20ms for the frame
length. Is this a logical explanation?

	ptime attribute of SDP is defined in RFC2327 and is obviously an
optional attribute. What do you think about implementing this option? Is
it possible that interop problems may arise due to the above phenomenon?

	Regards,

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