Voip server request to another VoIP servers

We have a requirement where a SIP request should send to a VoIP server and from that VoIP server should send a request to other peer VoIP servers ( One VoIP server request to multiple VoIP servers ). How do we acheive this ? How the configuration will be ? Thanks in advance

There is no team. This is is peer support forum.

Could you explain more about why you want to do this, as it is possible that you really want a SIP proxy, whereas Asterisk is a back to back user agent.

Also, which SIP method or methods are you referring to here, and for those that establish a session, what do you want to happen when one of the second level servers accepts the session.

Also why do you say to VoIP servers rather that SIP UAS. Asterisk can send to, for example, H.323 servers as well. Also, many people use server in a sense that is not used in the SIP specification. SIP doesn’t actually care whether the destination is a telephone or another PBX, so this doesn’t actually make much difference to Asterisk.

(The & operator in the Asterisk Dial application can be used to issue multiple parallel INVITEs and cancel the calls to those that lose the race. The ringall strategy for the Queue application can do similar. The Page application can issue multiple INVITEs and set up a one way audio stream towards all those that respond in time.)

Requirement :- When I play any audio from a Mobile App, it should play in an audio system in an Unit ( may be a room , office etc )

Here SIP client INVITE request will come from a Mobile device App. Once the SIP Invite sent to VoIP server , VOIP server will send an invite to the audio system ( MicroSIP for ex., ) . If audio system ( again a SIP user agent) is enabled, it will send response as 200 OK to VoIP server . VoIP server will send an OK message to Mobile device App and the device will play RTP streaming in the audio system ( loud speaker )

Now we have N number of units. Each unit will have its own VoIP server and Audio system ( Amon N audio systems one is considered as MASTER). Say for ex., From unit 1, the Mobile App request ( which is also inside Unit 1) sent to unit 1 VoIP server…THe VoIP server will then send request to Unit 1 Audio system , Unit 2 Audio system … Unit N Audio system…Whichever is the Master Audio system will respond OK to the VoIP server. Otherwise a DECLINE response to the VoIP server. The Mobile APp will then be played in the Master Audio system…

The first part sounds a bit like the Asterisk Page application, assuming the devices with the speakers can be configured to auto-answer.

However I got lost with the large number of servers mentioned the second part. Are these just the SIP UAS function that all SIP phones capable of receiving a request has, or are you talking about some intermediate device. Also SIP Page has no concept of a master device and I wasn’t clear whether the mater device refers to the device that can initiate the page, or a specific one of the paged devices (in the former case, it is just a matter of routing based on caller ID, etc.).

The Mobile Devices, which are the Voice Endpoints, interface via Wi-Fi provided. THere is no intermediate device here…Just a intermediate C++ code which hosts the VoIP server. which shall enable VoIP communication between the various End Points ( Speakers/Traditional phone etc).
For an analog based audio system , the Audio System Controller shall need to be configured / identified as a VoIP End Point and shall be responsible for the VoIP to Analog signal conversion and vice versa required for the actual physical End Points ( Speakers/Traditional phone )

Traditional, analogue, phones, won’t normally auto-answer calls.

THose are only to play the audio. Actual audio system is like a device again ( say MicroSIP )
The Audio System needs to be configured with the following information:
VoIP Server IP address/hostname,
SIP User Agent identity for each physical VoIP End Point ,
SIP User Agent password,
Codec Sampling rate,

To enable the Audio System to differentiate between the required functions, the Audio System shall need to host a separate SIP User Agent to represent each function

Simply I want an asterisk to asterisk communication

It’s hard to handle all SIP messages in Asterisk, you should use Kamailio to do this

Ok…Can I take this as there is no possibility to communicate one VoIP server to multiple VoIP servers and we use Kamailio ? ANy idea IAX will serve the purpose ?

You have basically the same restrictions with IAX as with SIP on Asterisk servers. Kamailio plus a C/C++ compiler is the way to go…

Asterisk is a modular system that deals with calls (SIP, IAX, DAHDI, …), whereas Kamailio is a SIP server that does not do anything beyond a SIP dialog, sometimes not even that. Asterisk offers Lego themes and occasionally some bricks, wheres Kamailio offers nuts, bolts, and occasionally some bricks (but none itself that do audio).

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