Is there still no response messages in the log with both reg_uri not set and public_addr set to your nat router public address when testing on private network. On 17/03/2013, at 4:47 PM, Ted Gerold <tedgerold at icloud.com> wrote: > > Same results using public_addr. tried it on both test systems. this is rough :) > > On Mar 16, 2013, at 10:30 PM, Ted Gerold <tedgerold at icloud.com> wrote: > >> the 207 was part of the tests from the public server. the 10.0 addresses were part of the tests from my private network >> server which I had to use to get the wireshark data. I have not come across public_addr before. I am looking into that now. >> >> >> >> On Mar 16, 2013, at 10:25 PM, Omar Hussein <omarh2812 at gmail.com> wrote: >> >>> 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 >> >> >> _______________________________________________ >> 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 >
