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.
Related
I want to make it easy for people to write bots for my ejabberd server by allowing them to register webhooks with their bot JIDs. Basically, I want them to be able to interact with my ejabberd server using HTTP to send and receive XMPP stanzas instead of having to maintain persistent XMPP connections. For example, the bots could simply make HTTP calls to my ejabberd server to send XMPP stanzas, and then the server would send messages to the bots by making HTTP calls to the registered webhooks. This would allow the bots to be simple HTTP servers that could easily scale horizontally.
I did find this module (https://github.com/adnam/ejabberd-webhooks) but it doesn’t seem to do exactly what I want and also it doesn’t seem to be maintained. Is there a well-maintained (maybe official) ejabberd module that does this? Maybe there is an XEP I’m not aware of that ejabberd implements? Are there other XMPP servers that support this? Thank you in advance!
the bots could simply make HTTP calls to my ejabberd server to send XMPP stanzas,
You can do this installing mod_rest in ejabberd:
https://github.com/processone/ejabberd-contrib/tree/master/mod_rest
and then the server would send messages to the bots by making HTTP calls to the registered webhooks.
I am not aware of any simple module to do this. There are Push XEPs, see https://xmpp.org/extensions/xep-0357.html but I think ejabberd Community Server does not implement everything required, only the Business Edition.
I am using a free caller android app and they restrict me with limited credits. I would like to explore more about how this app works. So I started decompiling the APK file and intercepting networking requests using Fiddler.
I intercepted requests, but none of those are creating the actual call. So after decompiling the APK I understood they are using SIP server for calls. I couldn't intercept any SIP call with fiddler or wireshark.
They are using this SIP server : https://www.kamailio.org/
Kamailio is one of the powerful SIP Servers and they have good documentation and big mail-user list. You can check their wiki:
http://www.kamailio.org/wiki/
For understanding this case first you should learn "how voip works" subject. You already can not inrecept any SIP call with fiddler or wireshark. Generally, SIP server managers hide their topology for security reasons. Thats why you can not observe SIP traffic to the end of the tunnel.
Even Kamailio has its own module that provides hide your network topology with one click :)
https://www.kamailio.org/wiki/tutorials/tls/testing-and-debugging#decoding_of_tls_connections_with_wireshark
If they can limit your credits, then, of course, they will authenticate your every call and reject it if you don't have enough funds (or out of quota).
So, if you learn a little bit about SIP technology - perhaps, you'll be able to send a call request even without a dialer, but I doubt that the call will be authenticated.
An analogy: if you know a full path to some restricted page - you can paste it into the browser's address field and go there, but if you are not logged in, then you will still get an error like "you're not allowed to read this page".
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.
I am implementing a VOIP application in which I work with SIP protocol. As per SIP I need to refresh my registration with SIP server at certain interval. But when my app goes in background, my keepalive handler is invoked only after 600 seconds as per documents os Apple. But this is not desired with SIP protocol. To be able to keep my connection alive with server and receive incoming call, I need to send registration message before 600 seconds even when app is in background. According to Apple documentation this is not possible but stilll I have seen apps on AppStore which runs in background and keeps their registration on with SIP server even when registration interval is 60 seconds. They keep app running in background throughout. So how is this possible? I know that playing silent audio in background will survive but then AppReview process will reject it. But if its so, how Apple allowed other such apps on AppStore? Is there some standard way to achieve what I have described above? Any help is appreciated.
You don't have to send the registration more often. You can do something like this: configure your server to send keep alive packages to your sip client in order to keep the TCP connection open. Your server should support TCP connection and your client should communicate over TCP since apple is very restrictive in background mode and only the TCP connection is allowed to remain open in the background (if your TCP socket is wrapped with CFReadStreamRef).
The problem is no the registration message, this can be configured by the client by specifying the time between the connection attempts, and the sip sever will be fine with an interval of one hour between 2 registration messages. The real problem is how will the server contact the sip client in case of an incoming call or IM. Most of the sip clients will not have a public IP adress but most probably they will be behind a NAT. So there is no way the sip server can open a direct connection to the sip client, for this reason you have to keep a socket open between your client and the server.
None of the app in the appstore is sending registration message more often than 600s.
Do you use your own library for sip or you use pjsip? If you use pjsip i can give you more hints. What SIP server do you plan to use?
Some hints:
make sure you set in your app-info.plist the required background modes to "App provides Voice over IP services" and "App plays audio"
in case your app will not be tied to your own sip server then provide a way for the user to disable the allow_contact_rewrite pjsip param (usually u want this to be enable for bypassing the nat problem) since some session border controllers are not happy about this feature
make sure, in case you capture the messages send form the sip thread in your main thread, you are using a method to post them in the main thread
here are described some other interesting issues
have a lot of coffee prepared :))
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..!