[Asterisk-Dev] Asterisk dies when bridging call

[Claus Futtrup]

I have this strange problem with Asterisk. Apparently Asterisk dies when
either recieving bad RTP trafic or when trying to bridge RTP streams between
a dead connection and a live one.

I've captured the flow with Ethereal and to my surprise there's alot of ICMP
destination unreachable.. digging into these I have found that the source
port and destination port from Asterisk matches what the called phone is
using. eg.

The SDP from Asterisk says that the phone should sent RTP to UDP port 48382
The SDP from the phoe says that Asterisk should sent RTP to UDP port 62459

After the SIP ack, the phone starts to sent audio to Asterisk from port
62459 to 48382 as stated in the SDP, but Asterisk does something rather
strange.
Asterisk sent from port 62459 to port 48382, and gets destination
unreachable. This eventually kills asterisk.(The packets are seen as ICMP
packets in Ethereal, not quite sure why)

Now the really strange thing is that this only occurs with one particular
phone, it works fine with everything else I have (xlite 2 xlite, xlite 2
zap, sjphone 2 xlite and so on).

Kind regards

Claus