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

Norman Franke norman at myasd.com
Mon Sep 10 14:23:16 EDT 2007


I'm using a bit dated version of pjsip (5.8 or so) and it works fine  
for me with Asterisk. I'd recommend monitoring the asterisk console  
(asterisk -r from the command line) and setting the verbosity to at  
least 3. You should then see the registration requests come in. I've  
noticed that Asterisk seems to require the following to match: realm  
(asterisk in my case), username and auth ID. I think it uses the auth  
id to lookup in sip.conf, and different clients handle auth id vs  
user name differently.

I have several pjsip clients authenticating to Asterisk in this  
manner and I've not encountered any problems. "sip show peers"  
reports in IP, port and latency (1 or 2ms in my case.)

-Norman

On Sep 10, 2007, at 9:57 PM, tech wrote:

> 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
>
>
>
> 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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>
>
> Call and stay connected with your friends and family for free. Seen  
> and be heard with high-definition video calls on Windows Live  
> Messenger. Try it!
>
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>
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