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milkman at the-milk-ba... Guest
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Posted: Sat Jan 12, 2008 7:11 pm Post subject: [asterisk-users] Packet2Packet bridging occurring when not w |
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Hi,
I have Asterisk set up on Fedora with a single SIP trunk, with a few
handsets configured. The Asterisk box has both public and private
addressing, so "canreinvite=no" is set on both the SIP trunk and handset
configurations so I can get around the nasty NAT issues.
One odd behaviour I am seeing is certain destinations are resulting in
different SIP codes being sent back to Asterisk, which then initiates
unwanted Packet2Packet bridging.
http://www.pastebin.ca/849961 <--- Working call. Line 280 shows a "SIP/2.0
183 Session Progress" and the RTP stream works as intended. I hung the call
up before being answered in case you were wondering where the answer part of
the debug occurs
http://www.pastebin.ca/849965 <--- Non-working call. Line 235 shows a
"SIP/2.0 180 Ringing", and for some odd reason, Packet2Packet bridging is
initiated despite "canreinvite=no" being set.
This in turn causes me some one way audio behaviour, and I've got no idea
how to fix it. Anyone able to offer any suggestions.
Regards,
Cameron Jenkins |
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joakimsen at gmail.com Guest
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Posted: Sat Jan 12, 2008 8:46 pm Post subject: [asterisk-users] Packet2Packet bridging occurring when not w |
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Packet2Packet has nothing to do with re-invite (well I guess if you
reinvite the RTP stream you can't use Packet2Packet on that box).
What Packet2Packet means is the RTP handler in Asterisk receives a
packet from say a SIP softphone and then sends that exact packet to
the SIP or IAX provider the call is going to without any additional
processing. This reduces the memory and CPU usage of Asterisk. See in
your own logs:
#
#
Sent RTP P2P packet to 210.8.38.129:13312 (type 08, len 000160)
#
Sent RTP P2P packet to 10.0.0.20:32120 (type 00, len 000160)
#
Sent RTP P2P packet to 210.8.38.129:13312 (type 08, len 000160)
#
Sent RTP P2P packet to 10.0.0.20:32120 (type 00, len 000160)
#
Sent RTP P2P packet to 210.8.38.129:13312 (type 08, len 000160)
#
Sent RTP P2P packet to 10.0.0.20:32120 (type 00, len 000160)
#
supermarket*CLI>
On Jan 12, 2008 7:11 PM, Cameron Jenkins <milkman at the-milk-bar.com> wrote:
Quote: | Hi,
I have Asterisk set up on Fedora with a single SIP trunk, with a few
handsets configured. The Asterisk box has both public and private
addressing, so "canreinvite=no" is set on both the SIP trunk and handset
configurations so I can get around the nasty NAT issues.
One odd behaviour I am seeing is certain destinations are resulting in
different SIP codes being sent back to Asterisk, which then initiates
unwanted Packet2Packet bridging.
http://www.pastebin.ca/849961 <--- Working call. Line 280 shows a "SIP/2.0
183 Session Progress" and the RTP stream works as intended. I hung the call
up before being answered in case you were wondering where the answer part of
the debug occurs
http://www.pastebin.ca/849965 <--- Non-working call. Line 235 shows a
"SIP/2.0 180 Ringing", and for some odd reason, Packet2Packet bridging is
initiated despite "canreinvite=no" being set.
This in turn causes me some one way audio behaviour, and I've got no idea
how to fix it. Anyone able to offer any suggestions.
Regards,
Cameron Jenkins
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joakimsen at gmail.com Guest
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Posted: Sat Jan 12, 2008 8:47 pm Post subject: [asterisk-users] Packet2Packet bridging occurring when not w |
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On Jan 12, 2008 7:11 PM, Cameron Jenkins <milkman at the-milk-bar.com> wrote:
Quote: |
This in turn causes me some one way audio behaviour, and I've got no idea
how to fix it. Anyone able to offer any suggestions.
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Seems like you are having a NAT issue. It would be nice if you posted
your exact network setup, what endpoints you are using and how they
are configured (including your SIP.conf) maybe someone could guide you
in the right direction if they knew what was going on. |
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