I have to integrate the text, voice and video chat via SIP server into my application. So that I have chosen the "Peers" from http://peers.sourceforge.net/.
I have downloaded the code, registered a sip addressz(peers sip client) and call to another sip account(peers sip client). I can't receive a call in that peers client. If I call to another sip client(X-Lite), I can able to receive a call.
Can anybody tell me what may be the problem and how to fix?
when you want to call a remote party from peers you should use sip:user#domain.com where domain.com is the domain you used in your account. If you used an ip address in domain in your sip account, you should use the same domain.
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How do sip addresses work? Is every email address a sip address? Like can I just ask two people for their email address and create a sip session between them? Can I issue sip addresses by creating emails for users? Thanks.
SIP address (more precisely: SIP URI) looks like an e-mail address but indeed it has totally different meaning. For example:
sip:john#example.com
means that there is an account named john on SIP Proxy server example.com (or on SIP Proxy server resolved from example.com).
When user john starts own SIP client, it sends a register message to SIP Proxy server. Thanks to this, the SIP Proxy server knows whether john is registered as well as his current IP address.
When other SIP user intends to create session with sip:john#example.com the following steps are performed:
example.com is translated into domain name of SIP Proxy server using DNS SRV.
The INVITE message is send towards this SIP Proxy server.
If john is registered, the INVITE is transferred to his client. If not, 404 response is sent back to originate user.
According your questions:
Is every email address a sip address? Definitely no.
Like can I just ask two people for their email address and create a sip session
between them? No.
Can I issue SIP addresses by creating emails for users? No.
We are a telecom company that allow our clients to call from a mobile application or their landline to friend or family around the globe...
Our system works like these application: https://play.google.com/store/apps/details?id=com.rebtel.android
https://play.google.com/store/apps/details?id=com.talkhome
We have three kind of calls:
1) - Direct call: A customer calls a virtual phone number with some dtmf code. This call is then transfered to a sip or voip call. The sip or voip then call the customer destination based on this dtmf code.
2) - Callback: A customer makes a web request via http connection or by sending an SMS to our platform. The sms contains some secret key, the destination number and some additional information. Base on these informations we will make request to our provider to make a call to the destination and our customer.
3) - Voip call: The client call through voip channel with his account.
We have already an termination voip provider. I'm looking for an company or a solution that can help me to connect from the DID numbers to the termination voip provider.
Do you have any idea?
Please clarify what's the missing part here?
Do You have SIP softswitch already? If yes let me know maybe i can help You in configuration to resolve the issue.
Am Using Pjsip libraries for SIP implementation . Using Pjsip lib i can able to make IP calls perfectly . My problem is While I Make call if other End User Decline/not Reachable or No answer ,My call will connect with Voice mail and get 200k(call connected state) from sip server as same like other end user Answering call.
Not Reachable Case i can differentiate 200 ok By missing 183 Session Progress Message. But noanswer and decline case I couldn't get the actual State of call .
is any way to find call connected state and Voice mail connected state ?
There isn't any one way of a user agent knowing it has reached a voicemail server versus a person answering.
Usually what will happen is that your user agent will send an INVITE to a From URI, and some proxy responsible for that URI will fork the INVITE to the various user agents the remote party uses: her phone, her voicemail service, and so on. The first user agent to return a 2xx response sets up the call.
You will however always know to which user agent you have connected, in the Contact header of the 200 OK.
Presumably a voicemail server will indicate in its SDP offer/answer that while it accepts certain incoming media streams, it will not send anything, by marking the stream with an a=recvonly attribute (See RFC 3264).
I don't know enough about SIP. As far as I know SIP can not be used in LAN. But it's features are very good. I want to use it for a LAN messenger (with video conference facilities).
Is there any way of using SIP in LAN network ?
The SIP protocol can be used over any reliable transport (TCP, XMPP, instant messaging channel, etc...) to a service (e.g. a server such as a SIP proxy) that knows how to route the SIP INVITE message from the caller to the callee. e.g. If you send an INVITE to bob#foobar.com, there's needs to be a service that knows how to find "bob" and deliver the message. Likewise, when Bob sends back his response messages back, the messages need to route back to the caller who sent the original INVITE.
And you can do SIP without a server - provided the computer already have a connection (direct or indirect) to the other computer intended for the call.
But SIP isn't anything special. If you were to invent your own video conferencing protocol, it would probably look a lot like SIP. SIP's primary job is for both sides of a call to exchange IP/port candidates for connecting directly in addition to codec and bandwidth negotiation data.
After the SIP messages are exchanged, ICE/STUN/TURN take over and RTP packets typically flow. SIP isn't used in the call except to end the call.
What are you really trying to do anyway?
Thread is Old but still I would like to contribute to this. There are various SIP server like http://www.officesip.com/index.html which works in LAN and can be connected to hardware phone too and soft client also.
Jitsi is open source cross platform SIP/xmpp client:https://jitsi.org/
And if you want to XMPP server Openfire is the best:http://www.igniterealtime.org/projects/openfire/
I hope this will definitely help someone..!
I am trying to make a gateway between SIP and XMPP domain. I have an environment with FreeSWITCH (with dingaling module) and OpenFire (FreeSWITCH is registered as a component in OF). Calls from XMPP to SIP are working fine. The problem appears for calls from SIP to XMPP. The caller (SIP) has a ringing tone, the callee (XMPP) has a 'connecting...' message after answering the call. The connection (voice channel) is never established.
Do you have any experience in making calls between SIP and XMPP users and have an idea, why it is not working?
Thanks in advance,
Maciek
Enable debug in mod_dingaling and mod_sofia and look for XMPP/Jingle errors.
However I suspect NAT/Firewall/Connections issues for this half-working behaviour.