Sorry for the late answer.
I looked into this issue this weekend, the SIP udp socket and the RTP udp
socket and we need to enable QoS for
RTP.
Our RTP stack builds upon libccrtp, which is based on the socket instance
defined in libcommoncpp2.
In libccrtp:
The RTPDataQueue contains:
typedef enum {
tosBestEffort, ///< Best-effort network service
tosEnhanced ///< Enhanced network service
}
and the coresponding method
inline void setTypeOfService(Tos tos) { typeOfService = tos; }
You may want to test setting this option in
/sflphone/sflphone-common/src/audio/audiortp/AudioRtpSession.h
Regards,
Alexandre
----- Original Message -----
From: "Thomas Alexander Ewald" <[email protected]>
To: "Alexandre Savard" <[email protected]>
Cc: [email protected]
Sent: Wednesday, October 27, 2010 7:48:01 AM GMT -05:00 US/Canada Eastern
Subject: Re: [SFLphone] Setting QoS bytes for SIP
Hi,
I've been messing around here with the code, but with no success.
*After* pjsip_udp_transport_start, I have added:
pj_status_t statQoS;
_debug("Getting socket");
pj_sock_t socket = pjsip_udp_transport_get_socket( transport );
statQoS = pj_sock_set_qos_type( socket, PJ_QOS_TYPE_VOICE );
if ( statQoS == PJ_SUCCESS ){
_debug("QoS type setted");
pj_qos_params qos_params;
statQoS = pj_sock_get_qos_params(socket, &qos_params);
if ( statQoS == PJ_SUCCESS ){
_debug("qos_params getted");
qos_params.flags = PJ_QOS_PARAM_HAS_DSCP;
qos_params.dscp_val = 0xB8;
_debug("Setting qos_params");
statQoS = pj_sock_set_qos_params(socket, &qos_params);
if ( statQoS == PJ_SUCCESS ){
_debug("Params setted");
}
}
}
Also, I have put this code after another transport search, in
method SIPVoIPLink::acquireTransport (const AccountID& accountID).
But Wireshark shows me that only packets from the server to my
machine have DSCP/TOS field value of 0xB8.
From my machine to the server nothing changes and the DSCP/TOS
field is 0x00.
I don't know exactly when I can set the QoS parameters. Before
pjsip_udp_transport_start the transport variable is NULL, so I cannot
get the socket, right?
Regards,
Thomas
On 10/26/2010 05:23 PM, Thomas Alexander Ewald wrote:
> Alexandre,
>
> Thanks for the help.
> Sure I'll test it here!
> For now I will try to activate QoS in a hacky and ugly way, just do
> see if it works.
> Are there any plans for a configuration option/interface?
>
> Again, thanks,
> Thomas
>
> On 10/26/2010 04:36 PM, Alexandre Savard wrote:
>> Hi,
>>
>> We are sometimes experiencing slow behavior on some networks so implementing
>> QoS
>> would be definitely a solution.
>>
>> We are using PJSIP implemented in
>> sflphone/sflphone-common/src/sip/sipvoiplink.cpp
>> The library is staticaly linked to sflphone from
>> sflphone/sflphone-common/libs/pjproject
>>
>> The unencripted udp sip transports are created in method:
>>
>> int SIPVoIPLink::createUdpTransport (AccountID id)
>>
>> using pjsip_udp_transport_start.
>>
>>
>> I found documentation describing activation of QoS with PJSIP using teh high
>> level API here http://trac.pjsip.org/repos/wiki/QoS . Unfortunately we are
>> dealing with the low level API.
>>
>> So we would need to acquire the internal socket from the transport using:
>>
>> pj_sock_t pjsip_udp_transport_get_socket( pjsip_transport *transport )
>>
>> And then activating the QoS type using the following API:
>>
>> http://www.pjsip.org/pjlib/docs/html/group__socket__qos.htm#gadbfe233355bf536b85ef39203d842715
>>
>>
>> I will take a look at it on my side. If you could test on yoour using your
>> setup that would be great!
>>
>> Regards,
>>
>> Alexandre
>>
>>
>> On 10/26/2010 02:46 PM, Thomas Alexander Ewald wrote:
>>> Hi,
>>>
>>> After struggling with my local SIP server, which only accepts
>>> clients with a fixed UserAgent ( and some SIP protocol changes too ), I
>>> have a new problem to solve.
>>> Now the server is QoS enabled, and the client has a configuration
>>> screen that allow to change the DSCP/TOS setting ( screen shot attached ).
>>> With this change, sflphone calls are extremely lagged, and the
>>> "authorized" clients are just fine.
>>> I search the code, but sincerely I'm lost.
>>> Where can I set the DSCP and TOS bytes for SIP packets?
>>> It can be hard coded, as I don't mind compiling all again ( anyway
>>> is what I actually do with every server change ) ;-)
>>>
>>> Thanks,
>>> Thomas
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