@paul_hadley wrote:
I read “please note that chan_sip will be going away in future versions of Asterisk” and felt a shudder go down my spine.
I had to replace my PBX last year and decided to build a purely pjsip system, as it was the future and I needed to get my head around the new channel it seemed an obvious choice.
Now I don’t claim to an expert but I can generally get my head around most IP PBX software. My first installations many years ago were Axon PBX, then Snom One, and more recently 3CX. In addition I have installed Grandstream PBX’s and Epygi systems, all without that much drama. I first started using FreePBX on a Raspberry PI about two years ago and have been installing FreePBX for about the last year. I would guess I have done more than 200 systems over the years and about 25 FreePBX systems to date all using chan_sip. So I am no expert with Asterisk but I am not a complete idiot with VOIP either.
From first use I was very impressed with FreePBX, moving into a Linux based systems from windows was always going to have its challenges but the transition was nowhere near as painful as I expected. The systems were easy to install, quite intuitive to set up and are feature rich, robust and very reliable, what was not to like.
Then my endeavours with pjsip started. Despite my best efforts I quickly had to abandon the idea of a pjsip only system, I simply could not get a system up and running, so I went for a compromise with chan_sip as the main channel and two extensions with two trunks on pjsip to test, evaluate and learn what was needed to get the channel working.
So far my problems with pjsip have been:-
- Pjsip trunks won’t update the IP address on remote servers when their dns moves to a new IP or the main server changes its IP. The problem is when one of the connections gets anew IP address; let’s say Server (A) gets a new external IP Address.
On server (A) all pjsip end points (extensions and trunks) become unavailable, phones cannot register with the extension status becoming unknown and the trunks show as unavailable.
They will not come back until I restart the asterisk service (service asterisk restart) when all is well again.At the same time server (B) will now show the trunk to server (A) as available again and allow outbound calls, however the "Match (Permit)" on the trunk does not update so no inbound calls are coming through.
Only a complete reload brings it all back to life.
I adjusted the trunk so that "Match (Permit)" was set to 0.0.0.0/0.0.0.0.0 to allow any connections, realising this might present a security issue it was only a temporary solution to test it. However once changed no extensions could register on the pjsip channel. As soon as I removed the entry all the extensions came back on line.
Another problem related to above appears be registering extensions that are mobile.
Two PBX’s connected with a pjsip for extension to extension dialling and working fine.
A soft phone (Linphone) installed on an iPhone and registering against PBX A and working fine. This can register anywhere in the world and works great except at the location of PBX B . When it tries to register from that location PBX A does not register it as an extension, it matches the external IP address to the trunk “Match: xxx.xxx.xx.xxx/32” section, tries to register it as a trunk not an extension and then the registration fails. This arrangement works fine with chan_sip.
- If the system becomes disconnected from the internet for even a few seconds then comes back online with the same IP on many occasions the pjsip channel is collapsing, all extension lose their registration and all trunks lose their connectivity.
Only a restart of the Asterisk brings it all back to life.
I have a Orbtalk (UK) trunk that uses an outbound proxy in its settings. I can get pjsip to register OK and make out bound calls fine, however on inbound calls the inbound audio fails, the caller can hear me but I can’t hear them, I haven’t been able to resolve this yet.
I have a Sipgate (UK) trunk, and again that will register fine and work on inbound calls, I get audio in both directions, but I am unable to make any out bound calls, the call just holds on the extension without ringing and after about 30 seconds I get a message saying the call was not answered. On the inbound calls the failure rate is about one in ten, I haven’t been able to resolve this yet.
I have a voipcheap (UK) account I only use for dialling out on. This appeared to set up fine and work, however about one in fifteen calls made on it fail, on chan_sip the failure rate is probably about one in five hundred.
Call pickup (*8) and (**) won’t work at all on pjsip extensions and chan_sip extensions cannot pick the calls up when the call is directed to a pjsip extension.
If I run pjsip on a PBX with more than one IP address I have to run the service on all the IP addresses, if I restrict the service to one IP address I cannot register any extensions or make calls between the trunks.
I am generally using Polycom phone and find the reboot feature in FreePBX very useful, however when the extension is changed over to pjsip the phone is not listed to be rebooted.
While adding video codecs to pjsip extensions does allow video to work fine I haven’t found any way to add video support to a pjsip trunk yet.
In general my efforts to implement pjsip have been a complete disaster. What started off as pjsip only system has now become (apart from one test trunk) a chan_sip only system.
I think a lot of the problems stem from using a system behind a dynamic IP Address so I am using dns to locate the system, this is working absolutely fine for chan_sip, and seems fully supported, but I do remember reading somewhere that it is not in pjsip. Clearly, when many small businesses and certainly home users don’t have access to reasonably price fixed IP address this may be quite a problem.
It should be noted that all the problems I have encountered with pjsip are on a system that is running chan_sip without any issues at all, so I am using a tested network, router, PBX Hardware and phones (including firmware). So in theory, if it all works with chan_sip but fails with pjsip then it points me to the problem being pjsip.
In general I would have to say that even when I have got pjsip working its feels very flakey and unstable compared with chan_sip. Chan_sip is not broken, but pjsip feels very broken, the problem is as end users we might not always have the option to choose. I really fear for the day I download FreePBX and find there is no chan_sip support.
I am not sure what other software uses the pjsip technology but I get the feeling it is better suited to supporting phones and soft phones than to pbx's and trunks, again maybe I have lost the plot, but at the moment I would rather see a future with chan_sip and not pjsip. If I saw an announcement saying pjsip in Asterisk had been abandoned then it would probably be the best IT news I had heard for a decade.
I don't know if other users have had better experiences than me with pjsip but when I have posed questions on this forum about its reliable implementation answers have been few and far between. Maybe I have just lost the plot, or maybe I am just one of many who have lost the plot !!!
To date, chan_sip I have implemented with ease, pjsip still has me completely beat !!
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