Just tested on one and the same PBX:
A: Softphone to softphone (MicroSIP and Zoiper): Almost no audio latency, maybe some 100ms.
B: Softphone to Digium phone (D80): Close to a full second, 1000ms! Someone can say one or two words before the signal arrives on the other end. That kind of latency is not good for the flow of a conversation.
Is there anything one can do about that?
Check your jitter buffer. If it is enabled, that can cause this without breaking a sweat.
That only exists in chan_sip, no?
IIRC, there is a jitter buffer in PJ-SIP, but I don’t know for sure. The trick is getting to know if it’s active in this case. Since you’re network is working well for your soft-phones, it would seem like something like that (at the extension level) might be getting in your way.
If you’re feeling brave, you could delete and rebuild the extension configuration to see if it makes any difference. If not, we’ll have to look somewhere else.
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