optimizing bandwidth usage for IVR applications writtenusing PJSIP

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Hi
 
Thanks a lot for your reply,
 
I think I figured out why I got twice the actual bit rate (e.g. 20KBytes/s
instead of 10KBytes/s for PCMU). I was using Windows Task Manager, Network
Tab, Bytes/Interval column to measure the bit rate. The refresh interval is
set to Normal which is every 2 seconds so I end up getting twice the actual
speed.
 
I will try to use G729 for a better bandwidth. But following your on your
calculations:
 
+ Using GSM codecs: 13kbps (payload) +16 kbps (RTP/UDP/IP headers)+2kbps
(SIP+RTCP) = 31kbps. This is quite close to my results.
+ Using G729 codes:  8kbps (payload) + 16kbps + 2kbps = 26kbps. 
 
So, compared to GSM, G729 only saves bandwidth by 5kbps, which is little. Is
my calculation wrong somewhere?
 
I have tried putting a call on hold using pjsua_call_set_hold. Calling this
the first time will eliminate the upload bandwidth (download bandwidth is
still there). Calling this the second time onwards will eliminate the
download bandwidth as well! However, any incoming DTMF is not detected
(which is what I wanted). 
 
Can you tell me in more details how I can do a sendonly in SDP? I guess you
mean PJMEDIA_DIR_ENCODING, which is available in pjsua_call_info.media_dir
but I am not sure how to set this for an existing call.
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