One thing that I don't understand is that in sip.log 18:10:13.837 pjsua_core.c SIP UDP socket reachable at 207.181.98.7:5060 18:10:13.837 udp0x1ee6ad0 SIP UDP transport started, published address is 207.181.98.7:5060 This shows that UDP transport is binding to 207.181.98.7:5060 but this is not your IP address of 10.0.1.10 (or was it at the time of testing in sip.log). The published address is what will be sent in the via header, contact header etc. This can be changed to the public IP address of your NAT router by setting public_addr in pjsua_transport_config struct. Regards, Omar -----Original Message----- From: Ted Gerold [mailto:tedgerold@xxxxxxxxxx] Sent: Sunday, 17 March 2013 8:26 AM To: Omar Hussein Cc: pjsip list Subject: Re: Problems with Outbound Calls No the soft phone is on a windows pc (10.0.1.4) and the test sip.c app is on a debian box (10.0.1.10). Both only have one NIC. I have 5060 forwarded to the debian box but no ports are forwarded to the windows box. On Mar 16, 2013, at 2:22 PM, Omar Hussein <omarh2812 at gmail.com> wrote: > So are you testing the soft phone from the same pc as pjsua? How many NICs are in the PCs and what is their ip configuration? > > > On 17/03/2013, at 7:35 AM, Ted Gerold <tedgerold at icloud.com> wrote: > >> So its easier to get the wireshark data via my local network (which >> I'm also using to test and getting the exact same results). Using my >> soft phone (which works great) I get the following from wireshark: >> >> 12 7.977004000 10.0.1.4 64.2.142.214 SIP/SDP 1081 Request: INVITE sip:6028675309 at outbound.vitelity.net;transport=UDP | , with session description >> 13 8.032728000 64.2.142.214 10.0.1.4 SIP 552 Status: 100 Trying | >> 14 8.849383000 64.2.142.214 10.0.1.4 SIP/SDP 895 Status: 183 Session Progress | , with session description >> 15 8.851479000 64.2.142.214 10.0.1.4 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x58AA0F24, Seq=12346, Time=2880 >> 16 8.851546000 10.0.1.4 64.2.142.214 ICMP 190 Destination unreachable (Port unreachable) >> 17 8.872092000 64.2.142.214 10.0.1.4 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x58AA0F24, Seq=12347, Time=3040 >> >> That looks good to me except #16. Every call I make that happens >> once and then it continues as normal. Maybe the issue with PJSUA is >> that it hangs on that part and doesn't know what to do. When I test >> using PJSUA there is a hang for a few moments after it starts the call before it proceeds to display logging output again. Course in PJSUA we aren't seeing the 100 and 183 codes so thats probably not it. >> >> This is very frustrating. :( >> >> >> >> On Mar 16, 2013, at 4:20 AM, Omar Hussein <omarh2812 at gmail.com> wrote: >> >>> Ok. I think it would be good to check the signalling of the working >>> softphone with wireshark. Then you will be able to see if the >>> softphone registers to the provider, what IP address it is sending >>> to etc. When you test the softphone is it from the same PC? >>> >>> It is just strange how the log does not show any response for the INVITE. >>> Normally you should see 100 Trying, 180 Ringing responses for >>> INVITE. The fact that the other phone rings means that you are >>> sending to the correct server. >>> >>> Your PC is listening for SIP messages on 207.181.98.7:5060. The via >>> header in the INVITE has this address so the provider should be >>> sending the response to this address. The address is a public IP >>> and so there is no NAT. >>> >>> Do you have a firewall blocking the response packets? Iptables etc. >>> >>> Regards, >>> Omar >>> >>> >>> >>> -----Original Message----- >>> From: Ted Gerold [mailto:tedgerold@xxxxxxxxxx] >>> Sent: Saturday, 16 March 2013 6:29 PM >>> To: Omar Hussein >>> Cc: 'pjsip list' >>> Subject: Re: Problems with Outbound Calls >>> >>> Same results :( >>> >>> >>> On Mar 16, 2013, at 12:12 AM, Omar Hussein <omarh2812 at gmail.com> wrote: >>> >>>> Hi Ted, >>>> >>>> If the provider does not require sending REGISTER before accepting >>>> call requests then don't set reg_uri in pjsua_acc_config struct. >>>> >>>> Regards, >>>> Omar >>>> >>>> >>>> -----Original Message----- >>>> From: Ted Gerold [mailto:tedgerold@xxxxxxxxxx] >>>> Sent: Saturday, 16 March 2013 3:44 PM >>>> To: pjsip list; Omar Hussein >>>> Subject: Re: Problems with Outbound Calls >>>> >>>> Hello Omar, >>>> Thanks so much for the reply. Regarding wireshark: Ive not heard >>>> of that I will have to look into it to see if I can use it to >>>> collect some more data. My provider does not require registration >>>> it works based on the IP address of my machine. >>>> The domain name they use is: outbound.vitelity.net however they >>>> list this on their site: >>>> >>>> Proxy: sip29.vitelity.net (66.241.99.28) Outbound Proxy: >>>> outbound.vitelity.net >>>> >>>> Since I am using the test sip.c app there is nothing in it that >>>> would allow me to test with any proxy. Is there code I could add >>>> to the account section that would accomplish the same thing? >>>> >>>> Thanks for the heads up on the password. It was just a test user >>>> so no big deal, easily deleted but should probably not be in there >>> regardless. >>>> >>>> New sip.c example if needed: >>>> >>>> http://dl.dropbox.com/u/61083309/www/content/sip.c >>>> >>>> -Ted >>>> >>>> >>>> On Mar 15, 2013, at 7:40 PM, Omar Hussein <omarh2812 at gmail.com> wrote: >>>> >>>>> Hi Ted, >>>>> >>>>> From sip.log it seems that there are no responses to REGISTER / INVITE >>>>> messages being sent to your provider. >>>>> >>>>> Since you have it working with a softphone perhaps a wireshark >>>>> trace of the working call compared to the non working call will help. >>>>> >>>>> Do you know if the provider requires Registration? What is the >>>>> domain of your account? Sometimes the domain (sent in To: header >>>>> etc. ) is separate to the server IP address you actually need to >>>>> send packets to. If that is the case you can configure the >>>>> --proxy setting in pjsua.exe sample application. >>>>> >>>>> P.S. I hope the username/password specified in your sip.c file are >>>>> not the real ones. :) >>>>> >>>>> Regards, >>>>> Omar >>>>> >>>>> >>>>> -----Original Message----- >>>>> From: pjsip [mailto:pjsip-bounces@xxxxxxxxxxxxxxx] On Behalf Of >>>>> Ted Gerold >>>>> Sent: Saturday, 16 March 2013 12:19 PM >>>>> To: PJSip >>>>> Subject: Problems with Outbound Calls >>>>> >>>>> Hello, >>>>> >>>>> I have spent a few days now trying to make an output call with >>>>> PJSAU but I can not seem to figure out why its not working properly. >>>>> >>>>> Here is the program I'm using (I got this example from some pj site): >>>>> >>>>> https://www.dropbox.com/s/ltic4hw5hm8p3xz/sip.c >>>>> >>>>> Here is my full log file: >>>>> >>>>> https://www.dropbox.com/s/61qtwoh25jox8d9/sip.log >>>>> >>>>> At first I was getting the 'cant find sound device' which I >>>>> expected as I'm not trying to connect to an external sound device. >>>>> I simply want to make a call, play a wav and hangup. So I used: >>>>> pjsua_set_null_snd_dev() to fix that. Now I can make the call and >>>>> the other end rings just fine but nothing happens after that. No >>>>> media state changes, no call_id given. It waits about 5 seconds >>>>> and disconnects (pjsua is not disconnecting as again there is no >>>>> media state >>>> change). >>>>> Then PJSUA just tries calling again. >>>>> >>>>> So to summarize: >>>>> >>>>> o call initiates >>>>> o i pick up phone and hear nothing o no media state change in >>>>> pjsua o pjsua calls me 2 more times before quitting. >>>>> >>>>> >>>>> At this point Ive probably spent at least 10 hours on this problem >>>>> and am desperate. Any help is much appreciated. I tried to >>>>> provide all possible information. >>>>> >>>>> Also I did test my DID provider Vitelity with a soft phone app and >>>>> it worked great. >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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 >>>>> >>>>> >>>>> _______________________________________________ >>>>> 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 >>
