26 Oct
2014
26 Oct
'14
11:20 p.m.
I fixed this. rtpengine must handle each of branches at branch_route(). Not
that is fine. Thanks for link. It was not my issue but with it i find right
way.
Can you help with these problem? We have 5-7 Seconds voice delay. This
happened only for from webphone. But it is not client issue as i see.
Wireshark at client side shows that RTP starts as soon I pick up call. So
rtp leaves rtpengine and goes to the destination with delay... We use WSS
and think that problem at handshake.
There are some statisticcs after call finished (as example). You may see
that one of streams created after 5 seconds delay.
P. S. Must we create new list for this?
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [
28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] --- Tag 'rsik48leli',
created 1:12 ago, in dialogue with 'as7b4cb593'
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [
28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34178
<> 8.2.10.25:52463, 340 p, 58032 b, 0 e
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [
28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34179
<> 8.2.10.25:52468 (RTCP), 10 p, 960 b, 0 e
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [
28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] --- Tag 'as7b4cb593',
created 1:12 ago, in dialogue with 'rsik48leli'
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [
28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34194
<> 10.0.1.6:16376, 201 p, 36582 b, 0 e
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [
28ed5c035d4b8a1c5383360e4d3677af@10.0.1.6:5060] ------ Media #1, port 34195
<> 10.0.1.6:16377 (RTCP), 1 p, 78 b, 0 e
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] Final
packet stats:
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] --- Tag
'as3af30098', created 1:17 ago, in dialogue with 'bqinihbhsf'
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------
Media #1, port 34142 <> 10.0.1.6:17258, 200 p, 35200 b, 0 e
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------
Media #1, port 34143 <> 10.0.1.6:17259 (RTCP), 1 p, 78 b, 0 e
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] --- Tag
'bqinihbhsf', created 1:12 ago, in dialogue with 'as3af30098'
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------
Media #1, port 34162 <> 8.2.10.25:52453, 216 p, 36768 b, 0 e
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [iunge05ber6sqlc7qs6b] ------
Media #1, port 34163 <> 8.2.10.25:52453 (RTCP),
2014-10-23 23:39 GMT+04:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>om>:
And what returns prtpengine at log when changing this
packet.
Returning to SIP proxy: d3:sdp316:v=0#015#012o=root 1195474335 1195474335
IN IP4 2.10.39.16#015#012s=Asterisk PBX 12.6.1#015#012c=IN IP4
2.10.39.16#015#012t=0 0#015#012m=audio 30614 RTP/AVP 8 3 0
101#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:3
GSM/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:101
telephone-event/8000#015#012a=fmtp:101
0-16#015#012a=ptime:20#015#012a=maxptime:150#015#012a=sendrecv#015#012a=rtcp:30615#015#0126:result2:oke
So it looks like that Destination sets from second append_branch at second
step (to UDP) and body sets as body of first step (for WS packet)
2014-10-23 23:36 GMT+04:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>om>:
No SDP body only one. but packet like this
INVITE sip:device-200@sip:1.21.10.2:45437;rinstance=07f88c423145358e;transport=UDP
SIP/2.0
Record-Route: <sip:sip.myservice.com:5068;nat=yes;ftag=as1be940e5;lr=on>
Via: SIP/2.0/UDP sip.myservice.com:5068
;branch=z9hG4bKca7d.2d16143316e23fac46bf686bb41780b3.2
Via: SIP/2.0/UDP 17.74.28.7:50600;branch=z9hG4bK22c67800;rport=50600
Max-Forwards: 70
From: "Name" <sip:1001@17.74.28.7:50600>;tag=as1be940e5
To: <sip:device-200@sip.myservice.com:5068>
Contact: <sip:1001@17.74.28.7:50600>
Call-ID: 5ee58acd136888261e85d91e345e7ba1@17.74.28.7:50600
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.6.1
Date: Thu, 23 Oct 2014 19:27:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 1044
v=0
o=root 1195474335 1195474335 IN IP4 2.10.39.16
s=Asterisk PBX 12.6.1
c=IN IP4 2.10.39.16
t=0 0
a=ice-lite
m=audio 30614 RTP/SAVPF 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp:30615
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:OY72ZDHa+E3avlHwschrdBMe00qDfkN0BUyOxT1C
a=setup:actpass
a=fingerprint:sha-1
07:3D:B4:B0:0E:0D:87:39:C3:83:10:E2:B8:B8:2C:0C:0D:59:EF:4C
a=ice-ufrag:Wudfwh08
a=ice-pwd:VoamuFVRrAXOhUaeD6tA3PcXhndL
a=candidate:8jYonvAy1KGkAdP3 1 UDP 213070
2014-10-23 23:25 GMT+04:00 Richard Fuchs <rfuchs(a)sipwise.com>om>:
On 10/23/14 15:06, Yuriy Gorlichenko wrote:
Still have same error...
Now rtpproxy_manage("co-sp") for classic call. At log I see that
rtpproxy wirked gine. For each step it generate write body, but t_Relay
still send strange "compinated" packet to UDP with SDP for WS...
Do you mean that the outgoing packet contains two SDP bodies? This has
been discussed and solved in this thread:
http://lists.sip-router.org/pipermail/sr-dev/2014-July/024507.html
cheers
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