Pjsip and sip

General Help
1

Can I have both types of extensions on the system at the same time? Thanks

2

Yes, you can mix and match any tech you want.

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3

Another stupid question - I can use a standard sip phone. I don’t need some sort of pjsip capable phone.

4

Chan_SIP and Chan_PJSIP are just SIP stacks. They are both SIP.

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5

Yes both really similar technologies. PJSip is designed to be better and easier to configure - and in theory it vastly is, however there are a couple of options in the ‘advanced’ fields that sometimes need to be filled in - if your provider or extension needs this, it’ll likely be asking for it in the logs :slight_smile:

6

Guys, there are multiple SIP stacks out there. FreeSwitch uses Sophia as do others, there’s SIP-B and others. Chan_SIP and Chan_PJSIP are just the SIP stacks that Asterisk uses.

SIP is SIP, it doesn’t matter what SIP stack is being used as long as it follows the SIP standards. Your providers doesn’t care what SIP stack you have on your PBX and your PBX doesn’t really care what SIP stack your provider is using. In that same vein, SIP phones are generally agnostic to the SIP stack as long as it is SIP.

7

The proviso is that the SIP stack you have chosen to use must support (and be configured to so support) all the SIP (Session Initiation Protocol) Initiations it receives from the Endpoint requesting them. :wink:

8

AKA “SIP Standards”

9

I am not disagreeing with you in any way, but if you look at the SIP RFC’s it would be hard to define such an “AKA”, if you restrict your view to just VOIP and perhaps video, then we are better positioned, but I would love to see a definitive “standard” for such :-), the balls in your court . . .

10

https://tools.ietf.org/html/rfc3261 <-- The one that explains the basics of SIP transactions and dialogs. The core of how INVITE, CANCEL, OPTIONS, REGISTER, etc, etc work. The core functions/tenants of what you expect out of SIP.

Example: INVITE is sent, if there is no 1XX reply within a time period it’s a timeout due to not responding. However, if there is a 1XX reply but it still times out, now it is a “No Answer” timeout because it means it reached the destination but no one answered in time and there was no voicemail, etc.

That is a standard of how that should work. ALL SIP stacks should handle that process the same way. Just like with REGISTERs and 401 auth challenges. These are common aspects of SIP that everyone learns and should be used by SIP stacks.Thus are the standard.

11

I agree , that is is how voip and SIP should work, and further down the SDP session needs to be negotiated and agreed on (think codecs) I was just saying that there is more to SIP than VOIP. (Yes I can be a pedant)

12

I may have only listed a few SIP methods that are related to calls. But I’m talking about SIP in general. The rules still apply to SIP MESSAGEs, PUBLISH, NOTIFY, SUBSCRIBES. There is standards for all of them that are followed.

Additionally, VoIP != SIP. SIP is a method of VoIP but H.323, IAX, SCCP and others like that are all considered part of VoIP. VoIP is just Voice over IP, it is not specific to a protocol.

But let us take a non voice aspect and compare. Let’s look at Presence and even more so Presence in Asterisk. Mainly due to the overall factor that Asterisk’s implementation of Presence is not considered part of the SIP standard in the RFC. Asterisk handles it differently and they even point that out as much. As much as I love Asterisk, let’s be clear that it is not 100% SIP standard compliant. There are ways in which Asterisk does things that those following standards have to adjust for when using Asterisk.

So again, SIP is a protocol and VoIP is a name given to the method of using voice over IP connections. Equating the two as equals should not happen.

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13

Who knew my questions was so loaded! So I setup a pjsip extensions. I am trying to get two phones to register to the same extension number. When a call is placed to extension 128 they both ring and both can answer with audio. So far so good. But only one phone can dial out or dial VM on the system. Any ideas?

14

Nevermind. I had one of the phones set to Encrypted for other tests. Everything seems to be working. The key was to set contacts to 2. Thanks everyone