Asterisk-User-Group.DE

Hallo,

ich werde mal die asterisk Version 12.0.0 testen und berichten

- neuer SIP Stack pjsip von http://www.pjsip.org
- added opus Audio Codec und VP8 Video Codec für webrtc http://www.webrtc.org
Pass through support has been added for both VP8 and Opus.
Added format attribute negotiation for the Opus codec. Format attribute negotiation is provided by the res_format_attr_opus module.
opus codec -> http://www.opus-codec.org/
VP8 codec -> http://www.webmproject.org/tools/

siehe -> http://downloads.asterisk.org/pub/telep ... Log-12.0.0
und -> https://www.youtube.com/user/asteriskvideos
Gruß
Rainer

Neu ist ...
man kann sich mit mehreren SIP Geräten an einem SIP Konto anmelden.
Die SIP Geräte klingeln dann parallel.

Gruß
Rainer

Neu ist ...
Asterisk Multithreading ...
Werde mal testen ob sich da was an der Performance verbessert hat.

wow ... pjsip soll 70000 normale SIP Packete pro Sekunde abarbeiten können.

Gruß
Rainer

ok ... erstes Problem libjansson-dev auf einem Debian squeeze fehlt.

Install libjansson-dev Howto:

Add the following line to /etc/apt/sources.list:

deb http://backports.debian.org/debian-backports squeeze-backports main

Update the package index:

# apt-get update

Install libjansson-dev deb package:

# apt-get install libjansson-dev

Gruß
Rainer

zweites Problem pjproject via git installieren.

# git clone https://github.com/asterisk/pjproject pjproject

# cd pjproject

# ./configure --prefix=/usr --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr
# make dep
# make
# make install
# ldconfig

Prüfen ob alles da ist mit ...
# ldconfig -p | grep libpj

Output sollte so aussehen ...
libpjsua.so (libc6,x86-64) => /usr/lib/libpjsua.so
libpjsip.so (libc6,x86-64) => /usr/lib/libpjsip.so
libpjsip-ua.so (libc6,x86-64) => /usr/lib/libpjsip-ua.so
libpjsip-simple.so (libc6,x86-64) => /usr/lib/libpjsip-simple.so
libpjnath.so (libc6,x86-64) => /usr/lib/libpjnath.so
libpjmedia.so (libc6,x86-64) => /usr/lib/libpjmedia.so
libpjmedia-videodev.so (libc6,x86-64) => /usr/lib/libpjmedia-videodev.so
libpjmedia-codec.so (libc6,x86-64) => /usr/lib/libpjmedia-codec.so
libpjmedia-audiodev.so (libc6,x86-64) => /usr/lib/libpjmedia-audiodev.so
libpjlib-util.so (libc6,x86-64) => /usr/lib/libpjlib-util.so
libpj.so (libc6,x86-64) => /usr/lib/libpj.so

Dann wieder in das Asterisk 12 Verzeichnis ...
## erforderliche libs installieren
# ./contrib/scripts/install_prereq install
# ./configure
# make menuselect
und unter Resource Modules sollten jetzt die res_pj*** Module aktivierbar sein.
# make
# make install
# make samples
# make config
# make install-logrotate

Gruß
Rainer

Hallo,

ich fahre zum testen die beiden SIP Stacks parallel.

Dafür muss man in der Datei pjsip.conf den Listen Port für pjsip hoch setzen.
Datei pjsip.conf

Code: Alles auswählen

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5072

Dann habe ich noch 2x PJSIP Konten angelegt.
Datei pjsip.conf

Code: Alles auswählen

[6001]
type = endpoint
transport = transport-udp
context = pjsipinternal
disallow = all
;allow = opus
allow = g722
allow = alaw
allow = ulaw
aors = 6001
auth = auth6001
direct_media = no
disable_direct_media_on_nat = yes
 
[6001]
type = aor
max_contacts = 10
 
[auth6001]
type=auth
auth_type=userpass
password=xxxxxxxx
username=6001
 
[6002]
type = endpoint
transport = transport-udp
context = pjsipinternal
disallow = all
;allow = opus
allow = g722
allow = alaw
allow = ulaw
aors = 6002
auth = auth6002
direct_media = no
disable_direct_media_on_nat = yes
 
[6002]
type = aor
max_contacts = 10

[auth6002]
type=auth
auth_type=userpass
password=xxxxxxxxxxxxxxx
username=6001

Datei extensions.conf

Code: Alles auswählen

[pjsipinternal]
exten => 6001,1,Dial(PJSIP/6001)
exten => 6002,1,Dial(PJSIP/6002)

Soweit alles OK

Timeout/Hangup nach 32 Sek.
Beide Endpoints sind nicht hinter NAT.

Problem CLI

Code: Alles auswählen

    -- Executing [7001@outgoing-kamailio:1] Dial("PJSIP/7000-00000002", "PJSIP/7001") in new stack
    -- Called PJSIP/7001
    -- PJSIP/7001-00000003 is ringing
    -- PJSIP/7001-00000003 answered PJSIP/7000-00000002
    -- Channel PJSIP/7000-00000002 joined 'simple_bridge' basic-bridge <74e9356e-73ee-451b-ab25-eb7cf125e9b5>
    -- Channel PJSIP/7001-00000003 joined 'simple_bridge' basic-bridge <74e9356e-73ee-451b-ab25-eb7cf125e9b5>
       > Bridge 74e9356e-73ee-451b-ab25-eb7cf125e9b5: switching from simple_bridge technology to native_rtp
[Dec 11 07:37:35] WARNING[28078]: pjsip:0 <?>:    tsx0x1f5af78 ...Temporary failure in sending Request msg INVITE/cseq=21539 (tdta0x2212530), will try next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
       > 0x7f962544e2f0 -- Probation passed - setting RTP source address to 192.168.8.10:6002
    -- Channel PJSIP/7001-00000003 left 'native_rtp' basic-bridge <74e9356e-73ee-451b-ab25-eb7cf125e9b5>
    -- Channel PJSIP/7000-00000002 left 'native_rtp' basic-bridge <74e9356e-73ee-451b-ab25-eb7cf125e9b5>
  == Spawn extension (outgoing-kamailio, 7001, 1) exited non-zero on 'PJSIP/7000-00000002'
    -- Executing [h@outgoing-kamailio:1] Hangup("PJSIP/7000-00000002", "") in new stack
  == Spawn extension (outgoing-kamailio, h, 1) exited non-zero on 'PJSIP/7000-00000002'

Work-a-round direct_media = no und disable_direct_media_on_nat = yes hinzufügen:

Code: Alles auswählen

[6001]
type = endpoint
transport = transport-udp
context = pjsipinternal
disallow = all
;allow = opus
allow = g722
allow = alaw
allow = ulaw
aors = 6001
auth = auth6001
direct_media = no
disable_direct_media_on_nat = yes

[6002]
type = endpoint
transport = transport-udp
context = pjsipinternal
disallow = all
;allow = opus
allow = g722
allow = alaw
allow = ulaw
aors = 6001
auth = auth6001
direct_media = no
disable_direct_media_on_nat = yes

Gruß
Rainer

Hallo,

habe jetzt die Release-Version 12.0.0 installiert.

ChangeLog zur Release-Version 12.0.0 siehe -> http://downloads.asterisk.org/pub/telep ... 12-current

Gruß
Rainer

Asterisk-User-Group.DE

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