[pjsip] Question : PJSIP client with Asterisk or OPENSER SIP server

tech tech at tge.com.sa
Mon Sep 10 21:57:31 EDT 2007


I had the same problem that Asterisk does not show registration info for my
pjsip client. However, the pjsip client can make/receive calls after I
initiate a call from pjsip (the first attempt fails, off course). Still
registration info does not show all the time.

 

Ahmad

 

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From: pjsip-bounces at lists.pjsip.org [mailto:pjsip-bounces at lists.pjsip.org]
On Behalf Of Nigel Hsiung
Sent: 10/Sep/2007 6:10 AM
To: Discussion forum for pjsip embedded/DSP SIP implementation
Subject: Re: [pjsip] Question : PJSIP client with Asterisk or OPENSER SIP
server

 

 


  _____  


Hi,

See comments below and hope it helps,

Nigel

Hi,
Currently I'm developing VOIP System.
I'm using PJSIP API to develop a SIP softphone on Windows Mobile 5.0. 
Before that, I'm using OPENSER SIP server as server, everything works fine
for registration, call... audio streaming.
For some reason, I need to change to asterisk as SIP server. When server
change the SIP client that developed from PJSIP works badly.

Here below is the description of structure:
- OPENSER/Asterisk is the SIP server that installed in linux behind  NAT A. 
- Desktop install X-lite  behind NAT A.
- Pocket PC install PJSIP SIP behind NAT B.



Following are problem i face during testing period:

Question 1:
1. Asterisk can be SIP server, SIP proxy?

Asterisk chan_sip can work as a (limited) sip server.  

2. Do I need OPENSER SIP server together with Asterisk. OR Asterisk  alone
will be fine?

Asterisk alone is fine unless u want to use TCP/TLS. 

Question 2:
Here is the question regard registering the account to Asterisk.
- several account has created under sip.conf.

For X-lite:
 1. registration account is fine,  when i check asterisk with command "sip
show peers" the host and port are listed with correct IP address and port.

For PJSIP SIP :
2. when registering an account, the account registered successfully with an
reply of 200 OK response message.
3. Yet when i check asterisk server with command "sip show peers" , the host
and port are UNDEFINED. As a result when i call other SIP account, it was
NOT FOUND message.

PJSIP is behind NAT B while Asterisk server is behind NAT A. Try PJSIP
behind NAT A and i think it'll work. Normally asterisk would not allow a
client to register from a different subnet. 

I don't have any ideal what course this issue....

Someone knows what happen around?
 Im' new to VOIP,  seeking some advice.

Thanks





 


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<http://www.get.live.com/messenger/features>  it! 

 

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