Call delay to call

[Adriano Vieira]

Hello, I would like to know if someone has already gone through this, when I make a call, through a SIP trunk, it takes 10 seconds to start calling .. is this delay normal?

When I test using asterisk, the call starts in less than 3 seconds.

Can someone help me?

LOG__________________________________________________________________________________________________________________________

2017-07-12 16:38:15.627560 [DEBUG] switch_core_state_machine.c:40 sofia/default/000005511941643630 Standard INIT

2017-07-12 16:38:15.627560 [DEBUG] switch_core_state_machine.c:48 (sofia/default/000005511941643630) State Change CS_INIT -> CS_ROUTING

2017-07-12 16:38:15.627560 [DEBUG] switch_core_state_machine.c:586 (sofia/default/000005511941643630) State INIT going to sleep

2017-07-12 16:38:15.627560 [DEBUG] switch_core_state_machine.c:543 (sofia/default/000005511941643630) Running State Change CS_ROUTING

2017-07-12 16:38:15.627560 [DEBUG] sofia.c:6858 Channel sofia/default/000005511941643630 entering state [calling][0]

2017-07-12 16:38:15.627560 [DEBUG] switch_core_state_machine.c:602 (sofia/default/000005511941643630) State ROUTING

2017-07-12 16:38:15.627560 [DEBUG] mod_sofia.c:142 sofia/default/000005511941643630 SOFIA ROUTING

2017-07-12 16:38:15.627560 [DEBUG] switch_ivr_originate.c:67 (sofia/default/000005511941643630) State Change CS_ROUTING -> CS_CONSUME_MEDIA

2017-07-12 16:38:15.627560 [DEBUG] switch_core_state_machine.c:602 (sofia/default/000005511941643630) State ROUTING going to sleep

2017-07-12 16:38:15.627560 [DEBUG] switch_core_state_machine.c:543 (sofia/default/000005511941643630) Running State Change CS_CONSUME_MEDIA

2017-07-12 16:38:15.627560 [DEBUG] switch_core_state_machine.c:621 (sofia/default/000005511941643630) State CONSUME_MEDIA

2017-07-12 16:38:15.627560 [DEBUG] switch_core_state_machine.c:621 (sofia/default/000005511941643630) State CONSUME_MEDIA going to sleep

2017-07-12 16:38:25.947465 [DEBUG] sofia.c:6858 Channel sofia/default/000005511941643630 entering state [proceeding][183]

2017-07-12 16:38:25.947465 [DEBUG] sofia.c:6868 Remote SDP:

v=0

o=CARRIER 1499877505 1499877505 IN IP4 XXX.XXX.XXX.XXX

s=SIP Call

c=IN IP4 xxx.xxx.xxx.xxx

t=0 0

m=audio 37482 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=ptime:20

LOG_____________________________________________________________________________________________________________________________________________

[Intel Networks]

You may need to define it abit more here.

Your log show negotiation as normal - when you testing at what point you getting a RBT? the delay can be that your endpoint (termination) is having RTP delay. Is your  endpoint running asterisk as well? From my experience is that asterisk and freeswitch seems to have some minute latency issues, i am not sure if it has to related to the core issue of the two call engine but what i will do if i were you is to adjust to get and early RBT that being said you will get a RBT as soon as the call initiated regardless whether it complete or not - if it doesn''t complete then thats when debugging started to find out the real problem. 

On Wed, Jul 12, 2017 at 12:42 PM, Adriano Vieira <gopl...@gmail.com> wrote:

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[Adriano Vieira]

How to adjust early RBT??

Tanks

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