A ramble about MSC, sip-connector and asterisk.

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Neels Hofmeyr nhofmeyr at sysmocom.de
Mon Jan 14 14:04:35 UTC 2019


Hi Gullik,

what is up with your email client? It seems to insert two line feeds every 100
characters or so. I've never seen such weird flow as in your mails. Harder to
follow your sentences when they break in the middle.

Using external PBX is not trivial, and as soon as you do, all local call
switching in osmo-msc is switched off.

In general, using osmo-sip-connector works; what is not satisfactory yet is the
codec negotiation. Sadly, so far the easiest solution is to hardcode a payload
type number into osmo-sip-connector and clamp everything to one specific codec.
For example, look at the neels/fr branch in osmo-sip-connector.git. But yours
sounds more like a SIP negotiation issue??

I haven't used asterisk yet, but AFAIK at least yate, freeswitch and kamailio
work without "clever hacking".

> when osmobts was bridging the calls

Nah. osmo-bts *never* bridges nor ever bridged calls.
Old openbsc's osmo-nitb? openbts??

> Boldly moving on to MultiBTS and handover....

Since you mentioned lime before, let me say that even using timing-accurate
sysmoBTS, I had 90% handover failure until I properly calibrated the internal
clock. With a lime and no external clock source, you're hopelessly doomed.

~N

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