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george.joseph at fairv...
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 Posted: Mon Feb 29, 2016 5:38 pm Post subject: [asterisk-users] PJSIP signaling question |
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On Mon, Feb 29, 2016 at 2:04 PM, Kevin Long <kevin.long@haloprivacy.com (kevin.long@haloprivacy.com)> wrote:
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Greetings.
I am using the PJSIP driver with TLS transport, and my endpoints are SIP mobile apps operating in environments that I do not control.
I would like Asterisk to default to sending INVITES and all other SIP signals to endpoints via the existing SIP TLS connection which is already established, rather than trying to create a new TLS connection to an endpoint which is likely behind a NAT which will not allow a new inbound TCP/TLS connection.
My experience with chan_sip suggest to me that this was the default behavior, or more likely a fallback behavior, because I never had this issue before with endpoints not receiving INVITES so long as they were registered and had an open SIP control connection.
I thought that I could avoid these failed outbound connections by commenting out the “transport” option on my endpoint configurations, but tcpdump is showing me that asterisk is still trying to create *new* TLS outbound connections to my endpoints, which are failing.
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This was actually an issue in pjproject which I just fixed last week. 
It's in pjproject "trunk" so you'll have to download and build it from their subversion repository.
Now whether you use "transport=" or not, pjproject will look for an existing connection to the remote endpoint before attempting to create a new one.
I tested it with the current Asterisk 13 branch and I *think* it'll work with recent Asterisk releases as well. If it doesn't, let me know.
| Quote: |
Thank you for your time
Kevin
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My simple pjsip config file:
[transport-tls]
type=transport
protocol=tls
bind=0.0.0.0:5061
local_net=10.50.55.0/24
external_media_address=x.x.x.x
external_signaling_address=x.x.x.x
cert_file=/etc/asterisk/keys/dev1.crt
priv_key_file=/etc/asterisk/keys/dev1.key
ca_list_file=/etc/asterisk/keys/ca.crt
cipher=AES256-SHA
method=tlsv1
;===============EXTENSION 6001
[6000]
type=endpoint
context=internal
disallow=all
allow=ulaw
;transport=transport-tls
auth=auth6000
aors=6000
direct_media=no
rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port
ice_support=no
force_rport=yes
rtp_symmetric=yes
media_encryption=sdes
[auth6000]
type=auth
auth_type=userpass
password=6000
username=6000
[6000]
type=aor
max_contacts=1
remove_existing=yes
;===============EXTENSION 6001
[6001]
type=endpoint
context=internal
disallow=all
allow=ulaw
;transport=transport-tls
auth=auth6001
aors=6001
direct_media=no
rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port
ice_support=no
force_rport=yes
rtp_symmetric=yes
media_encryption=sdes
[auth6001]
type=auth
auth_type=userpass
password=6001
username=6001
[6001]
type=aor
max_contacts=1
remove_existing=yes
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george.joseph at fairv...
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| Posted: Tue Mar 01, 2016 9:38 pm Post subject: [asterisk-users] PJSIP signaling question |
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On Tue, Mar 1, 2016 at 5:37 PM, Kevin Long <kevin.long@haloprivacy.com (kevin.long@haloprivacy.com)> wrote:
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Interesting, thanks George. I pulled Asterisk 13 from git and the new pjproject from the SVN and will test accordingly .
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Yeah, actually you do need Asterisk 13 from git because pjproject deprecated an api in trunk and we only handle that in the current git 13 branch.
| Quote: |
I have a few more questions about PJSIP in Asterisk 13:
1. Is there any way to list current ongoing calls and see what codecs are being used in the RTP streams? With chan_sip, “sip show channels” did this.
2. Also with a PJSIP initiated call, is there a way to see how man RTP packets have been sent and received for the call , I am debugging some intermittent 1-way and no-way audio on calls , and I am having trouble figuring out fi it is the client, firewall, or Asterisk/pjsip that is the culprit .
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Unfortunately, no to both (at least that I'm aware of). I remember looking at the channel stats a long while back and for some reason didn't go any further. I can re-look.
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george.joseph at fairv...
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| Posted: Fri Mar 04, 2016 3:01 am Post subject: [asterisk-users] PJSIP signaling question |
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On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.long@haloprivacy.com (kevin.long@haloprivacy.com)> wrote:
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Thanks George I appreciate the info . Being able to see what codec is in use for call in progress is very handy sometimes.
As far as the RTP stats goes, I see there is some info with “rtp” and “rtcp” commands which can be useful for troubleshooting. A running tally of # packets or bandwidth used would be awesome in along with the codec in "pjsip show channels" or something like that.
Im not certain, but I think the TLS signalling problem from this email may be happening to me again after patching for another pjsip/NAT issue which was with the external_media_address not working and the internal IP being sent in the SDP from asterisk - I applied this patch to the codebase and recompiled I am seeing the TLS “new transport” issue again , I think.
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I've lost track of who's applying what patches to which codebase.
Which patch did you apply for "external_media_address not working"?
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george.joseph at fairv...
Guest
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| Posted: Fri Mar 04, 2016 10:34 am Post subject: [asterisk-users] PJSIP signaling question |
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On Fri, Mar 4, 2016 at 1:16 AM, Kevin Long <kevin.long@haloprivacy.com (kevin.long@haloprivacy.com)> wrote:
| Quote: | Hi George the patch was from here , you wrote it I believe . I pulled asterisk 13 from git, apply this patch which fixed RTP issue , but I think tla transport issue came back for me .
https://gerrit.asterisk.org/#/c/2346/
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Oh, that one, OK. It should be merged now so if you 'git pull' on 13 now, you should get it. The transport re-use issue was in pjproject so is it possible that you're not compiling against the latest trunk?
| Quote: |
Thank you
Sent from my iPhone
On Mar 4, 2016, at 12:01 AM, George Joseph <george.joseph@fairview5.com (george.joseph@fairview5.com)> wrote:
| Quote: |
On Thu, Mar 3, 2016 at 8:25 PM, Kevin Long <kevin.long@haloprivacy.com (kevin.long@haloprivacy.com)> wrote:
| Quote: |
Thanks George I appreciate the info . Being able to see what codec is in use for call in progress is very handy sometimes.
As far as the RTP stats goes, I see there is some info with “rtp” and “rtcp” commands which can be useful for troubleshooting. A running tally of # packets or bandwidth used would be awesome in along with the codec in "pjsip show channels" or something like that.
Im not certain, but I think the TLS signalling problem from this email may be happening to me again after patching for another pjsip/NAT issue which was with the external_media_address not working and the internal IP being sent in the SDP from asterisk - I applied this patch to the codebase and recompiled I am seeing the TLS “new transport” issue again , I think.
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I've lost track of who's applying what patches to which codebase.
Which patch did you apply for "external_media_address not working"?
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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