il faudrait les traces (avec core set verbose 3 mini) de la console, et idéalement, les premières trames émises au début d'un appel avec rtp set debug on
il faudrait les traces (avec core set verbose 3 mini) de la console, et idéalement, les premières trames émises au début d'un appel avec rtp set debug on
Sécurisez votre asterisk, lisez ce post du forum: http://www.asterisk-france.org/showt...-recapitulatif et votre patton: http://www.asterisk-france.org/threa...tage-via-tiers - comprenez le nat : http://www.asterisk-france.org/threa...dio-pas-de-son
canreinvite est déprécié au profit de dircetmedia.
Plutôt que globalement, essayes directmedia = yes dans la config de tes clients et directmedia = no pour freephonie.
pour freephonie, ma config actuelle est plus simple.
Code:[freephonie](base_codecs) type=peer context=from-ext disallow=all allow=alaw,ulaw host=freephonie.net secret=xxxxxxxxxxx fromuser=09xxxxxxxx defaultuser=09xxxxxxx qualify=yes dtmfmode=inband fromdomain=freephonie.net insecure=port,invite canreinvite=no directmedia=no deny=0.0.0.0/0.0.0.0 permit=212.27.52.5/255.255.255.255Code:auth = 09xxxxxxxx:xxxxxxxxxxxx@freephonie.netCode:register => 09xxxxxxxx:xxxxxxxxxxxxxx@freephonie.net
L'utilisation d'une vm introduira une petite latence supplémentaire. Après cela dépend de la base matériel en premier, ensuite de l'hyperviseur utilisé. Je ne sait pas s'il existe des tests sur les performances "temps réel" des différentes solutions.
Voici les traces:
Code:========================================================================= Connected to Asterisk 13.7.0 currently running on srvasterisk01 (pid = 3272) == Using SIP RTP CoS mark 5 -- Executing [100@global:1] Dial("SIP/101-00000000", "SIP/100") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/100 -- SIP/100-00000001 is ringing -- Registered SIP '101' at 88.xxxxxxx:36996 > Saved useragent "Z 3.9.32144 r32121" for peer 101 [Feb 8 22:39:53] NOTICE[3779]: chan_sip.c:27701 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 101 [Feb 8 22:39:56] NOTICE[3779]: chan_sip.c:29374 sip_poke_noanswer: Peer '101' is now UNREACHABLE! Last qualify: 0 -- SIP/100-00000001 answered SIP/101-00000000 -- Channel SIP/100-00000001 joined 'simple_bridge' basic-bridge <801e5ab1-7e62-4589-bef5-3f58cf10efba> -- Channel SIP/101-00000000 joined 'simple_bridge' basic-bridge <801e5ab1-7e62-4589-bef5-3f58cf10efba> > 0x7f42b400bca0 -- Probation passed - setting RTP source address to 192.xxxxxx:8000 > 0x2002a00 -- Probation passed - setting RTP source address to 192.xxxxxx:8000 [Feb 8 22:40:06] NOTICE[3779]: chan_sip.c:23938 handle_response_peerpoke: Peer '101' is now Reachable. (5ms / 2000ms) [Feb 8 22:40:18] NOTICE[3779]: chan_sip.c:27701 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 101 [Feb 8 22:40:34] WARNING[3779]: chan_sip.c:4015 retrans_pkt: Retransmission timeout reached on transmission Y1gGaU5kx2gI9a_OTXTlSw.. for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 32001ms with no response [Feb 8 22:40:34] WARNING[3779]: chan_sip.c:4044 retrans_pkt: Hanging up call Y1gGaU5kx2gI9a_OTXTlSw.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). -- Channel SIP/101-00000000 left 'simple_bridge' basic-bridge <801e5ab1-7e62-4589-bef5-3f58cf10efba> == Spawn extension (global, 100, 1) exited non-zero on 'SIP/101-00000000' -- Channel SIP/100-00000001 left 'simple_bridge' basic-bridge <801e5ab1-7e62-4589-bef5-3f58cf10efba> srvasterisk01*CLI>
Ben la, c'est un problème réseau vers ton piste 101... Il passe brièvement offlibe et ne répond pas aux paquets de signalisation
Sécurisez votre asterisk, lisez ce post du forum: http://www.asterisk-france.org/showt...-recapitulatif et votre patton: http://www.asterisk-france.org/threa...tage-via-tiers - comprenez le nat : http://www.asterisk-france.org/threa...dio-pas-de-son
Je viens de modifier la conf et le problème réseau n'est plus en m'inspirant de la conf de olppp.. cependant j'ai toujours ce même problème de transmission de voix.
Code:[general] context=default srvlookup=no externip=88xxxxxxxxx localnet=192.xxxxxx/255.255.255.0 defaultexpirey=1800 dtmfmode=auto qualify=yes register => 09xxxxxx:xxxxxxx@freephonie.net language=fr ;disallow=all ;allow=ulaw allow=alaw ;allow=gsm qualify=yes canreinvite=no [freephonie] type=peer context=from-ext disallow=all allow=alaw,ulaw host=freephonie.net secret=xxxxxx fromuser=09xxxxxxx defaultuser=09xxxxxx qualify=yes dtmfmode=inband fromdomain=freephonie.net insecure=port,invite canreinvite=no directmedia=no deny=0.0.0.0/0.0.0.0 permit=212.27.52.5/255.255.255.255 register => 09xxxxxx:xxxxxxx@freephonie.net auth=register => 09xxxxxx:xxxxxx@freephonie.net [freephonie_inbound] type=peer host=freephonie.net context=global [freephonie_outbound] type=peer host=freephonie.net username= 09xxxxx secret= xxxxx fromuser= xxxxxxx fromdomain=freephonie.net [100] type=friend password=xxxx context=global host=dynamic callerid=accueil <100> [101] type=friend password= xxxxx context=global host=dynamic callerid=toto <101>Code:Using SIP RTP CoS mark 5 -- Executing [100@global:1] Dial("SIP/101-00000002", "SIP/100") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/100 -- SIP/100-00000003 is ringing > 0x1b42590 -- Probation passed - setting RTP source address to 192.168.0.4:8000 -- SIP/100-00000003 answered SIP/101-00000002 -- Channel SIP/100-00000003 joined 'simple_bridge' basic-bridge <ea2b54af-38eb-4403-9ac4-64a7823dc495> -- Channel SIP/101-00000002 joined 'simple_bridge' basic-bridge <ea2b54af-38eb-4403-9ac4-64a7823dc495> > 0x1b42590 -- Probation passed - setting RTP source address to 192.168.0.4:8000 > 0x7f5ec40080d0 -- Probation passed - setting RTP source address to 192.168.0.27:8000 -- Channel SIP/101-00000002 left 'simple_bridge' basic-bridge <ea2b54af-38eb-4403-9ac4-64a7823dc495> == Spawn extension (global, 100, 1) exited non-zero on 'SIP/101-00000002' -- Channel SIP/100-00000003 left 'simple_bridge' basic-bridge <ea2b54af-38eb-4403-9ac4-64a7823dc495>
Au passage dans sip.conf, section 100, il faut remplacer password par secret
Reteste avec rtp set debug on
Sécurisez votre asterisk, lisez ce post du forum: http://www.asterisk-france.org/showt...-recapitulatif et votre patton: http://www.asterisk-france.org/threa...tage-via-tiers - comprenez le nat : http://www.asterisk-france.org/threa...dio-pas-de-son
voici ce que j'obtiens jusqu'Ã ce que je raccroche sans son
Code:Using SIP RTP CoS mark 5 -- Executing [100@global:1] Dial("SIP/101-00000004", "SIP/100") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/100 -- SIP/100-00000005 is ringing -- SIP/100-00000005 answered SIP/101-00000004 -- Channel SIP/100-00000005 joined 'simple_bridge' basic-bridge <cb6de773-ee69-47d2-98a3-f3ad4c621633> -- Channel SIP/101-00000004 joined 'simple_bridge' basic-bridge <cb6de773-ee69-47d2-98a3-f3ad4c621633> > 0x1de4130 -- Probation passed - setting RTP source address to 192.xxxxx4:8000 Got RTP packet from 192.xxxxx.4:8000 (type 00, seq 031756, ts 796551540, len 000160) > 0x7fa06c00d7e0 -- Probation passed - setting RTP source address to 192xxxxx.27:8000 Got RTP packet from 192.xxxx.27:8000 (type 00, seq 042414, ts 2147281620, len 000160) Sent RTP packet to 88.xxxxx:8000 (type 00, seq 061833, ts 2147281616, len 000160) Got RTP packet from 192.xxxx:8000 (type 00, seq 031757, ts 796551700, len 000160) Got RTP packet from 192.xxx:8000 (type 00, seq 031758, ts 796551860, len 000160) Got RTP packet from 192.xxx.27:8000 (type 00, seq 042415, ts 2147281780, len 000160) Sent RTP packet to 88.xxxx:8000 (type 00, seq 061834, ts 2147281776, len 000160) Got RTP packet from 192xxxxx.27:8000 (type 00, seq 042416, ts 2147281940, len 000160) Sent RTP packet to 88.xxxxxx:8000 (type 00, seq 061835, ts 2147281936, len 000160) Got RTP packet from 192.xxxxx.4:8000 (type 00, seq 031759, ts 796552020, len 000160) Got RTP packet from 192.xxxxx.4:8000 (type 00, seq 031760, ts 796552180, len 000160) Got RTP packet from 192.xxxxx27:8000 (type 00, seq 042417, ts 2147282100, len 000160) Sent RTP packet to 88.xxxxxx:8000 (type 00, seq 061836, ts 2147282096, len 000160) Got RTP packet from 192.xxxxx4:8000 (type 00, seq 031761, ts 796552340, len 000160) ..... -- Channel SIP/101-00000004 left 'simple_bridge' basic-bridge <cb6de773-ee69-47d2-98a3-f3ad4c621633> == Spawn extension (global, 100, 1) exited non-zero on 'SIP/101-00000004' -- Channel SIP/100-00000005 left 'simple_bridge' basic-bridge <cb6de773-ee69-47d2-98a3-f3ad4c621633>