[asterisk-users] Unexpected delay

[flavio]

Hi to all.

I've registred my Asterisk 1.2.12.1 to a VoIP Service Provider and
I've some problem with outgoing calls: there is a big delay for
bidirectional audio flow.

Here is mean part of an asterisk trace releted to outgoing calls.
(canreinvite=no for both peers).
Until SIP 180 ringing signaling is correct...bold highlight time for NOTICE

_ _ _ _




Sep 18 16:01:43 [1;33;40mNOTICE[0;37;40m[23098]:
[1;37;40mchan_sip.c[0;37;40m:[1;37;40m9854[0;37;40m
[1;37;40mhandle_response_register[0;37;40m: Outbound Registration:
Expiry for 10.28.52.74 is 3599 sec (Scheduling reregistration in 3584
s)

[1;30;40m -- [0;37;40mSIP/outgoing-08197388 is ringing

Transmitting (no NAT) to 10.28.52.244:5060:
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP
10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244
From: <sip:bt102 at 10.28.52.246;user=phone>;tag=a82e9be13c882482
To: <sip:067202XXXX at 10.28.52.246;user=phone>;tag=as2ea0ddd1
Call-ID: a489c3f6ff15e77a at 10.28.52.244
CSeq: 829 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:067202XXXX at 10.28.52.246>
Content-Length: 0
---

[1;30;40m -- [0;37;40mSIP/outgoing-08197388 is making progress passing
it to SIP/bt102-08190d90
Sep 18 16:02:37 [1;33;40mNOTICE[0;37;40m[23098]:
[1;37;40mchan_sip.c[0;37;40m:[1;37;40m11613[0;37;40m
[1;37;40msip_poke_noanswer[0;37;40m: Peer 'outgoing' is now
UNREACHABLE! Last qualify: 4


<-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

<-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

<-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

<-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

<-- SIP read from 10.28.52.244:5060:

--- (0 headers 0 lines) Nat keepalive ---

[1;30;40m -- [0;37;40mSIP/outgoing-08197388 answered SIP/bt102-08190d90

We're at 10.28.52.246 port 16274

Adding codec 0x4 (ulaw) to SDP

Adding codec 0x8 (alaw) to SDP

Reliably Transmitting (no NAT) to 10.28.52.244:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
10.28.52.244;branch=z9hG4bKc39fdf160cf737ac;received=10.28.52.244
From: <sip:bt102 at 10.28.52.246;user=phone>;tag=a82e9be13c882482
To: <sip:067202XXXX at 10.28.52.246;user=phone>;tag=as2ea0ddd1
Call-ID: a489c3f6ff15e77a at 10.28.52.244
CSeq: 829 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:067202XXXX at 10.28.52.246>
Content-Type: application/sdp
Content-Length: 184
v=0
o=root 23109 23110 IN IP4 10.28.52.246
s=session
c=IN IP4 10.28.52.246
t=0 0
m=audio 16274 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -

---

[1;30;40m -- [0;37;40mAttempting native bridge of SIP/bt102-08190d90
and SIP/outgoing-08197388
Sep 18 16:03:25 [1;33;40mNOTICE[0;37;40m[23098]:
[1;37;40mchan_sip.c[0;37;40m:[1;37;40m9882[0;37;40m
[1;37;40mhandle_response_peerpoke[0;37;40m: Peer 'outgoing' is now
REACHABLE! (6ms / 2000ms)

<-- SIP read from 10.28.52.244:5060:
ACK sip:067202XXXX at 10.28.52.246 SIP/2.0
Via: SIP/2.0/UDP 10.28.52.244;branch=z9hG4bK30db550457acdb99
From: <sip:bt102 at 10.28.52.246;user=phone>;tag=a82e9be13c882482
To: <sip:067202XXXX28.52.246;user=phone>;tag=as2ea0ddd1
Contact: <sip:bt102 at 10.28.52.244;user=phone>
Call-ID: a489c3f6ff15e77a at 10.28.52.244
CSeq: 829 ACK
User-Agent: Grandstream BT110 1.0.8.12
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0

_ _ _ _

>From trace it points out that time gap from 180 Ringing and follow 200
Ok is about 1 minute.. and so from 200 OK and ACK
Any suggestions?

Moreover..when I attempt to make an outgoing call with option
canreinvite=yes, Asterisk notifies the follow message?

Sep 20 14:13:42 WARNING[2373]: channel.c:787 channel_find_locked:
Avoided initial deadlock for '0x819b240', 10 retries!

Can anyone tell me what it does mean and how to fix it?


Thanks 4 all


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