[SR-Users] rtpengine not working with append_branch

Sun Oct 26 23:20:52 CET 2014

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 at 10.0.1.6:5060] --- Tag 'rsik48leli',
created 1:12 ago, in dialogue with 'as7b4cb593'
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [
28ed5c035d4b8a1c5383360e4d3677af at 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 at 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 at 10.0.1.6:5060] --- Tag 'as7b4cb593',
created 1:12 ago, in dialogue with 'rsik48leli'
Oct 26 22:15:31 Kamailio2 rtpengine[35466]: [
28ed5c035d4b8a1c5383360e4d3677af at 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 at 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 at gmail.com>:

> 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 at gmail.com>:
>
>> No SDP body only one. but packet like this
>>
>> INVITE sip:device-200 at 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 at 17.74.28.7:50600>;tag=as1be940e5
>> To: <sip:device-200 at sip.myservice.com:5068>
>> Contact: <sip:1001 at 17.74.28.7:50600>
>> Call-ID: 5ee58acd136888261e85d91e345e7ba1 at 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 at sipwise.com>:
>>
>>> 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
>>>
>>> _______________________________________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>>> sr-users at lists.sip-router.org
>>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>
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