2007/9/10, Nigel Hsiung <nigelcz at hotmail.com>: > > > > ________________________________ > 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. Asterisk is a SIP B2BUA.. it does not act as a SIP Proxy.. but do you really need of a SIP Proxy? :-) > > 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. You are really sure that client receives 200 OK or are you sure that Asterisk sends 200 OK SIP message? Tipically where host and port are UNDEFINED it is because UAC is not correctly registrered.. tipically.. :D > > 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. Asterisk allows nat if you configure it in sip.conf. reed more at http://www.voip-info.org/wiki-Asterisk+sip+nat :-) > > I don't have any ideal what course this issue.... > > Someone knows what happen around? > Im' new to VOIP, seeking some advice. > > Thanks I hope this may be helpfull > > > > > > ________________________________ > 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! > ________________________________ > 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! > ________________________________ > Get news, entertainment and everything you care about at Live.com. Check it > out! > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > -- ******************************** * (o< ing.Flavio Patria * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************
