Subject: [Sipping] Draft of SIP Telephony Device Requirements

From: "Miao, Kai" <[email-masked]>
Date: Fri, 18 Apr 2003 09:05:03 -0700

Henry,

Sorry for the big delay for (re)sending these comments on the SIP device
drafts to this list: 
 
1) SIP devices with an analog user front-end probably need to 
be singled out in someway. For example, the document 
mentioned the T.38 only applies to this type of SIP devices, 
but does not include other differences with them such as echo 
cancellation, tone detection, tone generation, etc.
 
2) In "IP requirements", it does not mentioned ICMP, nor IGMP 
amd multicast. These probably should be at least "may" items?

3) This document only mentions DTMFs and ringback for tone 
support. There can be other tones which may also be required, 
such as call progress tones transferred over via RFC2833 when 
a call comes from a IP-PSTN gateway, dial tone when a user 
first "picks up" the device, etc. In addition, tones may also 
be transported "out of band" via signaling channel (using, 
for example,
SIP INFO method), has this been intentionally left out?     
 
4) For multi-line support, it does not mention if a call or 
even a conference can be established between the lines. I 
like the local mixing requirement for two or more parties 
from the network. But it is not clear to me how such mixing 
capability relates to the local multiple lines that are available.

5) I didn't any see volume control and level control 
mentioned anywhere. Is something like ALC (G.169) required 
for SIP devices. 
 
6) The silence suppression related statements in the document 
do not seem to be clear to me. RFC 3389 specifies for RTP 
comfort noise payload format only for codecs like G.711, 
G.726, G.722, etc. Exactly the same applies to G.711 Annex 
II. For codecs like G.729 and G.723.1, CNG/VAD is internally 
defined (see RFC1890).

7) I am also wondering why there is no mention of anything 
related to delay variation (netwrk jitter) compensention. Is 
PLC (G.711 annex I) reqquired? 
 
8) SIP security is mentioned in the document, not media 
security. Is SRTP required?
 
Best Regards,
Kai

_______________________________________________
Sipping mailing list  https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use [email-masked] for questions on current sip
Use [email-masked] for new developments of core SIP