Hi,
I am using Asterisk 1.6.2.6 at server side. PJSIP version 1.8 at client side.
I have custom board which work as SIP phone which communicate with Soft-phone installed in Windows PC via Asterisk server.
Each SIP phones connected to network switch(we called it as member switch) so by Ethernet cable board & member switch connected to each other.
All member switches connected to Main network switch so network related issues solved by it. Server & soft-phone also connected to Main network switch.
But due to Power consumption issue, we decide that member switch should go to sleep mode till event not occurs.
For R n D we do following test:-
- Every thing is up & running properly. In Asterisk ‘sip show peers’ shows properly.
- stop the member switch(we simple off the power of that network switch). & look how much time takes to unregistered from Asterisk.
- wait for time t after unregistered event comes.(This waiting time t, we use 2,5,15 minutes etc)
- After waiting time t complete, start member switch.
- find out after how much time in Asterisk we can see sip phone registered.
For observation, i keep log level at 10.
Observation-
- When we wait for 2 min after unregistered, for Registration Asterisk-PJSIP takes 3 min 46 sec approx.
- When we wait for 5 min after unregistered, for Registration Asterisk-PJSIP takes 36-40 sec approx.
- When we wait for 15 mins after unregistered, for Registration Asterisk-PJSIP takes 1 min 50 sec approx.
For reduce this Registration time we do following changes & take above Observation:-
- As per ‘PJSIP-Dev-Guide.pdf’ page 13, we changes in PJSIP’s ‘sip_endpoint.c’ at Line no. 698.
Original code:-
/* timer_heap_poll should never ever returns negative value, or otherwise
* ioqueue_poll() will block forever!
*/
pj_assert(timeout.sec >= 0 && timeout.msec >= 0);
if (timeout.msec >= 1000) timeout.msec = 999;My changes code:-
/* timer_heap_poll should never ever returns negative value, or otherwise
* ioqueue_poll() will block forever!
*/
pj_assert(timeout.sec >= 0 && timeout.msec >= 0);
if (timeout.msec >= 1) timeout.msec = 1;following is result of ‘sip show settings’:-
Global Settings:
----------------
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: No
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: No
Direct RTP setup: No
User Agent: Asterisk PBX 1.6.2.6
SDP Session Name: Asterisk PBX 1.6.2.6
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
Externip: 0.0.0.0:0
Externrefresh: 10
Internal IP: 10.111.17.3:5060
STUN server: 0.0.0.0:0
Global Signalling Settings:
---------------------------
Codecs: 0x8100e (gsm|ulaw|alaw|g722|h263)
Codec Order: g722:20
Relax DTMF: No
RFC2833 Compensation: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
----following result of ‘sip show peer 0066’:-
* Name : 0066
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : users
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : Yes
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : 61
Insecure : port,invite
Nat : RFC3581
ACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : Yes
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : No
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost :
Addr->IP : 10.111.17.66 Port 5060
Defaddr->IP : 0.0.0.0 Port 5060
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 0066
SIP Options : (none)
Codecs : 0x8100e (gsm|ulaw|alaw|g722|h263)
Codec Order : (g722:20)
Auto-Framing : No
100 on REG : No
Status : Unmonitored
Useragent : PJSUA v1.8/arm-none-linux-gnueabi
Reg. Contact : sip:0066@10.111.17.66:5060
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
Parkinglot :I tried to change in ‘Sess-Expires’ & ‘Min-Sess’ in sip.conf but there is no change in observation!
Question:-
- At where i should make change so after network switch turn on, then in very less time PJSIP register at Asterisk?
Need help.