hi all,
i am having problem related to ptime.
platform: I am having ARM platform on client side and Asterisk as PBX
server
I am offering * PJSUA_DEFAULT_AUDIO_FRAME_PTIME = 10*
(pjsip/include/pjsua-lib/pjsua.h)
and* G722 codec @ 16khz* and *--auto-answer enabled*
I am giving you list of conference ports before call establishment*
*
Buddy list:
-none-
+=============================================================================+
| Call Commands: | Buddy, IM & Presence: |
Account: |
| |
| |
| m Make new call | +b Add new buddy .| +a Add new
accnt |
| M Make multiple calls | -b Delete buddy | -a Delete
accnt. |
| a Answer call | i Send IM | !a Modify
accnt. |
| h Hangup call (ha=all) | s Subscribe presence | rr
(Re-)register |
| H Hold call | u Unsubscribe presence | ru
Unregister |
| v re-inVite (release hold) | t ToGgle Online status | > Cycle next
ac.|
| U send UPDATE | T Set online status | < Cycle prev
ac.|
| ],[ Select next/prev call
+--------------------------+-------------------+
| x Xfer call | Media Commands: | Status &
Config: |
| X Xfer with Replaces |
| |
| # Send RFC 2833 DTMF | cl List ports | d Dump
status |
| * Send DTMF with INFO | cc Connect port | dd Dump
detailed |
| dq Dump curr. call quality | cd Disconnect port | dc Dump
config |
| | V Adjust audio Volume | f Save
config |
| S Send arbitrary REQUEST | Cp Codec priorities | f Save
config |
+------------------------------+--------------------------+-------------------+
| q QUIT L ReLoad sleep MS echo [0|1|txt] n: detect NAT
type |
+=============================================================================+
You have 0 active call
>>> cl
Conference ports:
Port #00[16KHz/*10ms*/1] Master/sound transmitting to:
Port #01[16KHz/*10ms*/1] scomb-rev transmitting to:
*and when server send 200 OK responce to client it offers ptime= 10 ms
but at the time of call establishment it established with ptime = 20ms
I am also giving packet of 200 OK responce from server where it shows ptime
= 10 ms
*
>>> 14:19:31.319 pjsua_core.c RX 834 bytes Response msg
200/INVITE/cseq=11461 (rdata0x15545c) from UDP 192.168.0.3:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.10:5060
;branch=z9hG4bKPjlqmd7OMi96C5Y98RUL5aJa0t4Vh8SxH1;received=192.168.0.10;rport=5060
From: sip:0080@192.168.0.3;tag=z3eX0XyxqyMBjjNy8bnIL.bpC7emvgLl
To: sip:0050 at 192.168.0.3;tag=as31f911b3
Call-ID: E6.i7JuHv8aSFFXI8KfKhoduJn6nzyts
CSeq: 11461 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uac
Contact: <sip:0050 at 192.168.0.3>
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 767109880 767109880 IN IP4 192.168.0.3
s=Asterisk PBX 1.6.2.6
c=IN IP4 192.168.0.3
t=0 0
m=audio 12738 RTP/AVP 9 101
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
*a=ptime:10*
a=sendrecv
--end msg--
*and list of conference ports after establishing call*
You have 1 active call
Current call id=1 to sip:0050 at 192.168.0.3 [CONFIRMED]
>>> cl
Conference ports:
Port #00[16KHz/10ms/1] Master/sound transmitting to:
Port #01[16KHz/10ms/1] scomb-rev transmitting to:
Port #02[16KHz*/20ms*/1] sip:0050 at 192.168.0.3 transmitting to:
*I want to know is why pjsua taken ptime as 20ms since server offered pjsua
ptime=10ms??????*
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