Chan_sip vs pjsip

[Marbled (Nicolas Riendeau)]

Hi,

@ashleydrees

To be honest, it amazes me that if you have both channel sip and chan pjsip on the same system FreePBX hijacks the SIP port for chan pjsip - even though in ALL the docs from FreePBX they state that it is not stable - IF they are going to run both on the same setup - one would presume that the stable one would be on the standard port and the flaky/beta/indevelopment/dont use because it will break - thing on the “alternative” port -

Here we go again…

Some light reading for you…

See

PJSIP still has bugs

Don’t use PJSIP yet is is very buggy Mod edit: this was taken from New installation reload failed still

and

Is PJSIP stable?

Do all the features of of pbxact work great with pjsip or should I stick with chan_sip? Is there a reference guide on what the benefits are for PJSIP over chan_sip? Thank you in advance

Have a nice day,

Nick

[makadonkie]

Hey jamied_uk,

I presume you may have already come to an answer, but I found that even changing the SIP port on my Cisco IP Phone made no difference to the port that it was attempting to use. Instead, when you define the proxy, add the port number to the end, ie; 192.168.1.200:5160

Hope this works for you!

[Solako]

@paul_hadley

I have another trunk with orbtalk (UK Supplier) that requires an out bound proxy, that also registers fine with pjsip, calls can be made in and out but I only get audio in, no outbound audio on external calls.

Lol, This sounds like a quagmire I have been trying to get myself out of.

[Solako]

@paul_hadley

My biggest problem has been dealing with dynamic IP addresses. With chan_sip a change of IP at any location can quickly be recovered from with all trunks, service providers and internal links back up and working in less than five minutes without any need for user intervention.

However on pjsip this has proved a nightmare. Firstly the on pbx behind the IP change the pjsip channel collapses completely, no trunks are available and interestingly even pjsip extensions become unavailable and only recover after restarting the asterisk service.

Trrruuuuueee! hahahahaha. Quite a learning curve.

[Solako]

@paul_hadley

and if the pbx is behind a router with a dynamic IP address pjsip is not a viable option at the moment.

hey @paul_hadley how did yo configure the router to allow voice calls out/in?

[etoven (Erich Toven)]

5 years later PJSIP stil breaks my trunk. Not touching it.

[BlazeStudios (Tom Ray)]

You have no choice at this point. Chan_sip is dead.

[PitzKey (Itzik)]

I’m not aware of anything in PJSIP that would break an endpoint connection. Even the tel URI is now supported… I suggest creating a post with your setup and issue.

[petecat1 (Peter Arnold-Nott)]

There are definitely numbers that if we call using a pjsip trunk, there are one way audio issues, especially 1800 numbers with IVR receiving the call (in Australia). SIP trunk works for those every time.

[BlazeStudios (Tom Ray)]

Have you done actual troubleshooting?

[jcolp (Joshua Colp)]

The underlying media stack is identical between chan_sip and chan_pjsip. The only thing different in that regard is the SDP negotiation itself, or different configuration.

[petecat1 (Peter Arnold-Nott)]

I haven’t troubleshot it other than to route calls to 1800 numbers over the trunk with the ChanSIP configuration which fixed the issue and that was good enough for me. I figure the problem lies in the setup at the other end. There are a small number of individual numbers this often happens on as well, and problem is always fixed by using the ChanSIP outgoing trunk.

[BlazeStudios (Tom Ray)]

Well chan_sip is dead and doesn’t exist in any version of Asterisk beyond v20. Not sure how much support it is going to get for the next couple years before 20 goes EOL.

If you’re having a problem that you believe is equated to chan_pjsip, I’d suggest you do a little deeper troubleshooting as if the issue is with chan_pjsip it will most likely be a configuration issue some where.

[Stewart1 (Stewart)]

@petecat1

I haven’t troubleshot it other than to route calls to 1800 numbers over the trunk with the ChanSIP configuration …

This sounds strange. If it’s the same account with the same provider, why don’t you have only the chan_sip trunk, used for all calls? (I’m definitely not recommending that, just want to understand the logic.)

In the problematic case with pjsip, did you determine that there is no outbound audio, rather than a DTMF issue? For example, you called a larger company with speech-enabled IVR and speaking the option also didn’t work?

[petecat1 (Peter Arnold-Nott)]

I have two providers set up for outgoing calls. I prefer provider 1 due to cost. Thats the one with pjsip that’s set up for most outgoing calls. its also the one that doesn’t work for 1800 numbers. So I force calls to 1800 numbers to the other trunk, provider 2, which is set up using chan_sip. I guess I could set up a pjsip trunk using provider 2, and send the 1800 calls out through that to test if its the provider and not the pjsip. But from memory I had issues with pjsip on outgoing calls when I first set this up and that provider 2 was my only provider. (I could also set up a chan-sip trunk with provider 1 and send the 1800 numbers over that to also test if its the provider. I realise this is against the ethos of pure research, but honestly, if its working…)

In the problematic case with pjsip, we get through the IVR ok (using DTMF), and when finally got through to a person, we could hear them but they could not hear us.

[petecat1 (Peter Arnold-Nott)]

Then I read BlazeStudio’s comment.
I’ll have a look when I get some time.

[BlazeStudios (Tom Ray)]

Post the chan_sip trunk config and the chan_pjsip trunk config. We can compare the settings and see what might be different.

[jcolp (Joshua Colp)]

This thread on the Asterisk side came to mind: Ignore SDP Version - Asterisk SIP - Asterisk Community

[petecat1 (Peter Arnold-Nott)]

Just to feedback. I’ve just tried my sip trunk and my pjsip trunk to the 1800 and the 13 numbers and all are working fine. I’ll leave it on the pjsip for a while and see what happens. I will hear pretty quickly if folks are getting audio problems.