platform: I am having ARM platform on client side and Asterisk as PBX

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*

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@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@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@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@192.168.0.3 transmitting to: *I want to know is why pjsua taken ptime as 20ms since server offered pjsua ptime=10ms??????*