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Hi,
Current SIP proxy never modifeis RTP, so what comes in mind:
1) If proxy has natted public IP, you need to fill host name or at leat put public IP there.
2) You havent specified STUN server in SIP client.
What sip client you used, cureentyl i use X-lite, hardware Siemens SL75 WLAN, Linksys SPA 3102 pstn gateway.
Please be free to ask more, i'm willing to debug out whats problem.
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Thanks for your answer!
The proxy server as no NAT, it has a regular public IP (207.179.xxx.xxx).
As client, I have tried with 2 x-lite (I did not specified the STUN server because they autodetect it.)
I tried with diferent IP phones too (Linksys, Sippura, Cisco, Grandstream)... they all works with Asterisk, SER, LCS but not with your Proxy.
PS: if you want me to run some interoperatibility test for you for the future, I have plenty of SIP Hardware, Servers with Public IPs, and SIP <-> PSTN Gateway to test with.
Yann
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X- won't autodedect STUN, you need to specify it. Like stun.counterpath.net or stunserver.org.
Also if you set Host NAme in proxy to domain name, or if no domain name, use public IP, this
ensures ACK command reaches UA1 to UA2. Probably if you do both(specify stun and host name) all works as you expect.
Same goes for phones.
Why asterisk works, probably you run it B2BUA mode and RTP goes through it.
What NAT router do you have ? SO far i haven't meet any what won't work.
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OK,
The person with who I did the test had a wrong stttings on his Router... I fixed it and it all works fine now.....
Thank you,
Yann
PS: Do you plan any time soon on adding some RTP/RTCP on it? to act as B2BUA or more like Asterisk? I was planing on doing it with Directshow maybe... what do you think?
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I'm currently doing b2bua, but with out RTP. Most of times RTP proxying not good idea, it consumes twice of your bandwidth, adds delay to call, ... . RTP proxy is needed only if you would to spy/record calls, othwerwise there is no use.
About testing, what you offered, all tests and feedback are welcome.
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Of course, RTP proxy is not always a good idea but it's some case a RTP stack is needed (Conference bridge, Voicemail, auto attendant,...). I will start doing some test with Directshow... did you ever try?
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No i don't use Directshow, if i complete b2bua and have nothing other todo, i code RTP stack in plain C#. Then add simple codecs like g711 and some other not licensed ones.
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I was thinking about writing the all RTP stack in C# but did not know if c# will be fast enought for Real Time video/voice compare to directshow that is writed in C...? but I guess it could works fine.... I will run some test when I get a change...
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> if c# will be fast enought for Real Time video/voice
I'm sure that speed difference isn't very big. Also CPU power will gorw, Microsoft does good work for improving C# speed, ... - i dont see any reason not to use C#.
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Nice Work again....... what's next? RTP or Presence?
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Have you ever had this working with Microsoft Live Communications server as a call control gateway?
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Hi,
No, i just once installed Microsoft Live Communications server 2007 and looked fast what it does.
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I download your sip demo from your web page. I try to do some test using the X-lite as SIP client. The test consist in a communication between two PC. I can register the two client but can not establish a call between then. I appreciate your help. If then problem have a solution I think that the Sip solution is very good.
Other question do yo thinks in a call between PC - PSTN ...
thank for all
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Hi,
Redownload it from my www, i updated i, that version must definately work - i use it.
About pstn, it under development, i just got my SIP - pstn gateway (linksys spa3102), so during some days, it will be done.
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Could you please summarize again purpose and functionality? I cannot see this working as a real proxy. It does REGISTERs, ok, hence it is somewhat like a registrar. But a proxy? I doubt that... Is it able to forward unknown URIs to another proxy?
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Hi,
This is full proxy (only not a calls statefull what can control call).
Latest support:
Registrar
Statless proxy
Statefull proxy
No Forking
Sequential Forking
Prallel Forking
----
Recurse - redirect call if redirect response got.
Currently i use hardware phone "siemens sl 75 wlan" and soft phones with it.
If you have some troubles, you can email me directly.
I'll soon update codeproject version too, if all tests done.
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Thanks for the quick answer. Is it possible to perform outbound calls (e.g. using another public registrar/proxy in order to forward calls, which cannot be resolved within the local domain?)
Regards
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Offcourse, thats the main job of proxy.
This is complete phone system app, except call statefull what allows to manage call state and it doesn't have RTP(voice/video) proxy.
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OK, but where to provide the credentials of the forgein registrar/proxy?
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thats the job of SIP phone application.
Proxy just forward auth request back to user.
(Even foreign ones, then SIP message has multiple auth request headers and
SIP phone must ask username/pwd for each of them)
normally you never need to auth to remote server, why ?
because local proxy calls right destination server what is final or at least responsible
remote domain. If domain is proxy local,normally no auth asked or you can't call to "this" user.
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Hi,
will this SIP Proxy be able to do what Asterisk or SER Proxy Servers are capable of doing? Or is it only coapable of doing a subset of those works?
Can I use this proxy server to make a call from my client Soft phone application to a PSTN?? i.e., any other device(mobile/landline) or any other phone(VOIP Device) ?
Can I make use of RTC API of Microsoft with this?
Rgds,
Srikanth
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Hi,
Asterisk or SER is more advanced, but it does most important features what is needed.
>Can I use this proxy server to make a call from my client Soft phone application to a PSTN??
Yes, but you need hardware gateway or then some PSTN service provider. PSTN Gateway just must be listed in geteways, like uriScheme=tel, ip of gateway, .... .
>or any other phone(VOIP Device) ?
Yes.
>Can I make use of RTC API of Microsoft with this?
It depends what you want to do, for simeple sip, yes.
This server won't implement b2bua, but if know right SER doesn't too, but most of times it isn't needed (why to pass all audio/video trough your server).
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Hi,
again I have some basic questions regarding SIP servers. In our company we already are making use of VOIP Phones (Hardware phones provided by Nortel) .. and we are making use of VOIP services (I suppose), coz thats the only thing which enables us to make VOIP calls isn't it?(Apart from servers like Asterisk).... so I assume that we have a ITSP (internet Telephony service providers just making sure that I know the abbreviation properly .. ) ......
So assuming the above condition, do I need to use a SIP Proxy server?? wudn't the ITSPs provide a SIP server??
If you think that we may already be making use of ITSPs SIP server, do you think that I can make use of just RTC API 1.3 and code the SIP client application for that?
Also, could you please guide me about how to go about developing the Client SIP Application for the web applications?? Is it the same the way we develop for Desktop applications like Softphones?
Thank you in advance.
Regards,
Srikanth
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>I assume that we have a ITSP (internet Telephony service providers
I don't get exactly what you mean. You can run own server for all, no service provider needed at all. Just for VOIP you need SIP proxy, for pstn calls you can but SIP -> PSTN gateway like what i used Linksys spa 3102 or better one if you need more PSTN lines.
>do you think that I can make use of just RTC API 1.3 and code the SIP client
Yes, you can use RTC API.
>Is it the same the way we develop for Desktop applications like Softphones?
You need activeX or java applet for that, you can't carry voice otherwise.
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