One way call

PL
Peter Lukac
Tue, Apr 13, 2010 2:25 PM

hello all
I have special request. I want create call where RTP stream will be sent only
one way from caller to callee. Not from callee to caller.

  1. exists some "method in SIP message" how can i say collee to not sent RTP
    stream? Exists something like user data in SIP message? If yes how can i set
    it?

  2. how can i disable RTP stream on collee side? (some API) ..respectively on
    UA side.I found callback on_stream_created() it is good place? :)

thanks

hello all I have special request. I want create call where RTP stream will be sent only one way from caller to callee. Not from callee to caller. 1) exists some "method in SIP message" how can i say collee to not sent RTP stream? Exists something like user data in SIP message? If yes how can i set it? 2) how can i disable RTP stream on collee side? (some API) ..respectively on UA side.I found callback on_stream_created() it is good place? :) thanks
AO
Alejandro Orellana
Tue, Apr 13, 2010 2:47 PM

look at the sdp attributes

a=sendonly, rcvonly sendrcv

http://www.ietf.org/rfc/rfc3264.txt

Cheers

On Apr 13, 2010, at 10:25 AM, Peter Lukac wrote:

hello all
I have special request. I want create call where RTP stream will be sent only one way from caller to callee. Not from callee to caller.

  1. exists some "method in SIP message" how can i say collee to not sent RTP stream? Exists something like user data in SIP message? If yes how can i set it?

  2. how can i disable RTP stream on collee side? (some API) ..respectively on UA side.I found callback on_stream_created() it is good place? :)

thanks


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Alejandro Orellana
Savant Systems, LLC
770 Main Street
Osterville, Ma. 02655
(508) 683-2547 (W)
(508) 683-2647 (F)
Alejandro.Orellana@savantav.com

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look at the sdp attributes a=sendonly, rcvonly sendrcv http://www.ietf.org/rfc/rfc3264.txt Cheers On Apr 13, 2010, at 10:25 AM, Peter Lukac wrote: > hello all > I have special request. I want create call where RTP stream will be sent only one way from caller to callee. Not from callee to caller. > > 1) exists some "method in SIP message" how can i say collee to not sent RTP stream? Exists something like user data in SIP message? If yes how can i set it? > > 2) how can i disable RTP stream on collee side? (some API) ..respectively on UA side.I found callback on_stream_created() it is good place? :) > > > thanks > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org ------ Alejandro Orellana Savant Systems, LLC 770 Main Street Osterville, Ma. 02655 (508) 683-2547 (W) (508) 683-2647 (F) Alejandro.Orellana@savantav.com Statement of Confidentiality The contents of this e-mail message and any attachments are confidential and are intended solely for the addressee. The information may also be legally privileged. This transmission is sent in trust, and the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply e-mail or at 508.683-2500 and delete this message and its attachments, if any.
PL
Peter Lukac
Wed, Apr 14, 2010 6:26 AM

thanks for advice. I look at it.

look at the sdp attributes

a=sendonly, rcvonly sendrcv

http://www.ietf.org/rfc/rfc3264.txt

Cheers

On Apr 13, 2010, at 10:25 AM, Peter Lukac wrote:

hello all
I have special request. I want create call where RTP stream will be sent
only one way from caller to callee. Not from callee to caller.

  1. exists some "method in SIP message" how can i say collee to not sent
    RTP stream? Exists something like user data in SIP message? If yes how
    can i set it?

  2. how can i disable RTP stream on collee side? (some API) ..respectively
    on UA side.I found callback on_stream_created() it is good place? :)

thanks

thanks for advice. I look at it. > look at the sdp attributes > > a=sendonly, rcvonly sendrcv > > http://www.ietf.org/rfc/rfc3264.txt > > > Cheers > > On Apr 13, 2010, at 10:25 AM, Peter Lukac wrote: > > hello all > > I have special request. I want create call where RTP stream will be sent > > only one way from caller to callee. Not from callee to caller. > > > > 1) exists some "method in SIP message" how can i say collee to not sent > > RTP stream? Exists something like user data in SIP message? If yes how > > can i set it? > > > > 2) how can i disable RTP stream on collee side? (some API) ..respectively > > on UA side.I found callback on_stream_created() it is good place? :) > > > > > > thanks > >
PL
Peter Lukac
Mon, Apr 26, 2010 12:23 PM

hello again

How can i add/modify DPS protocol in SIP message when i creating call?

in pjsua_call_make_call method is parameter msg_data..

pj_status_t pjsua_call_make_call
(
pjsua_acc_id
acc_id,

const pj_str_t *
dst_uri,

unsigned
options,

void *
user_data,

const pjsua_msg_data *
msg_data,

pjsua_call_id *
p_call_id

)

and i suppose this parameter i have to adjust. But how?
when i try:

pjsua_msg_data msg_data;

pjsua_msg_data_init( &msg_data );

pj_str_t hname = pj_str("application/sdp");
pj_str_t hvalue = pj_str("a=sendonly");

msg_data.content_type  = hname;
msg_data.msg_body  = hvalue;

pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id);

in SIP message is still:

"a=sendrecv"

struct "pjsua_msg_data" have another member  "hdr_list" of type  "pjsip_hdr"
Do i have to adjust this member too?..if yes how?

thanks

hello again How can i add/modify DPS protocol in SIP message when i creating call? in pjsua_call_make_call method is parameter msg_data.. pj_status_t pjsua_call_make_call ( pjsua_acc_id acc_id, const pj_str_t * dst_uri, unsigned options, void * user_data, const pjsua_msg_data * msg_data, pjsua_call_id * p_call_id ) and i suppose this parameter i have to adjust. But how? when i try: pjsua_msg_data msg_data; pjsua_msg_data_init( &msg_data ); pj_str_t hname = pj_str("application/sdp"); pj_str_t hvalue = pj_str("a=sendonly"); msg_data.content_type = hname; msg_data.msg_body = hvalue; pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id); in SIP message is still: "a=sendrecv" struct "pjsua_msg_data" have another member "hdr_list" of type "pjsip_hdr" Do i have to adjust this member too?..if yes how? thanks
NI
Nanang Izzuddin
Tue, Apr 27, 2010 6:29 AM

Hi Peter,

On pjsua level, this may be easily done using this trick:

  1. normal make call, pjsua_call_make_call()
  2. once the call gets confirmed, put the call on-hold,
    pjsua_call_set_hold(), call hold will put 'sendonly' direction
    attribute in the sdp.

BR,
nanang

On Mon, Apr 26, 2010 at 7:23 PM, Peter Lukac p.lukac@emtest.sk wrote:

hello again

How can i add/modify DPS protocol in SIP message when i creating call?

in pjsua_call_make_call method is parameter msg_data..

pj_status_t pjsua_call_make_call (pjsua_acc_id acc_id,
const pj_str_t * dst_uri,
unsigned options,
void * user_data,
const pjsua_msg_data * msg_data,
pjsua_call_id * p_call_id
)
and i suppose this parameter i have to adjust. But how?
when i try:

pjsua_msg_data msg_data;

pjsua_msg_data_init( &msg_data );

pj_str_t hname = pj_str("application/sdp");
pj_str_t hvalue = pj_str("a=sendonly");

msg_data.content_type = hname;
msg_data.msg_body = hvalue;

pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id);

in SIP message is still:

"a=sendrecv"

struct "pjsua_msg_data" have another member "hdr_list" of type "pjsip_hdr"
Do i have to adjust this member too?..if yes how?

thanks


Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

Hi Peter, On pjsua level, this may be easily done using this trick: 1. normal make call, pjsua_call_make_call() 2. once the call gets confirmed, put the call on-hold, pjsua_call_set_hold(), call hold will put 'sendonly' direction attribute in the sdp. BR, nanang On Mon, Apr 26, 2010 at 7:23 PM, Peter Lukac <p.lukac@emtest.sk> wrote: > > hello again > > How can i add/modify DPS protocol in SIP message when i creating call? > > in pjsua_call_make_call method is parameter msg_data.. > > > pj_status_t pjsua_call_make_call (pjsua_acc_id acc_id, > const pj_str_t * dst_uri, > unsigned options, > void * user_data, > const pjsua_msg_data * msg_data, > pjsua_call_id * p_call_id > ) > and i suppose this parameter i have to adjust. But how? > when i try: > > pjsua_msg_data msg_data; > > pjsua_msg_data_init( &msg_data ); > > pj_str_t hname = pj_str("application/sdp"); > pj_str_t hvalue = pj_str("a=sendonly"); > > msg_data.content_type = hname; > msg_data.msg_body = hvalue; > > pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id); > > in SIP message is still: > > "a=sendrecv" > > struct "pjsua_msg_data" have another member "hdr_list" of type "pjsip_hdr" > Do i have to adjust this member too?..if yes how? > > > thanks > > > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >
PL
Peter Lukac
Tue, Apr 27, 2010 11:32 AM

hi

thanks for reply.. this trick is little hack :) and i preffer clean
sollutions...it is posible with struct "pjsua_msg_data"? or some callback? or
some another way? ( use pjsua API is not necessary  but welcome :) )

I want control rtp stream all directions (sendonly,recvonly and sendrecv).

thanks

Hi Peter,

On pjsua level, this may be easily done using this trick:

  1. normal make call, pjsua_call_make_call()
  2. once the call gets confirmed, put the call on-hold,
    pjsua_call_set_hold(), call hold will put 'sendonly' direction
    attribute in the sdp.

BR,
nanang

On Mon, Apr 26, 2010 at 7:23 PM, Peter Lukac p.lukac@emtest.sk wrote:

hello again

How can i add/modify DPS protocol in SIP message when i creating call?

in pjsua_call_make_call method is parameter msg_data..

pj_status_t pjsua_call_make_call (pjsua_acc_id acc_id,
const pj_str_t * dst_uri,
unsigned options,
void * user_data,
const pjsua_msg_data * msg_data,
pjsua_call_id * p_call_id
)
and i suppose this parameter i have to adjust. But how?
when i try:

pjsua_msg_data msg_data;

pjsua_msg_data_init( &msg_data );

pj_str_t hname = pj_str("application/sdp");
pj_str_t hvalue = pj_str("a=sendonly");

msg_data.content_type = hname;
msg_data.msg_body = hvalue;

pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id);

in SIP message is still:

"a=sendrecv"

struct "pjsua_msg_data" have another member "hdr_list" of type
"pjsip_hdr" Do i have to adjust this member too?..if yes how?

thanks


Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

hi thanks for reply.. this trick is little hack :) and i preffer clean sollutions...it is posible with struct "pjsua_msg_data"? or some callback? or some another way? ( use pjsua API is not necessary but welcome :) ) I want control rtp stream all directions (sendonly,recvonly and sendrecv). thanks > Hi Peter, > > On pjsua level, this may be easily done using this trick: > 1. normal make call, pjsua_call_make_call() > 2. once the call gets confirmed, put the call on-hold, > pjsua_call_set_hold(), call hold will put 'sendonly' direction > attribute in the sdp. > > BR, > nanang > > On Mon, Apr 26, 2010 at 7:23 PM, Peter Lukac <p.lukac@emtest.sk> wrote: > > hello again > > > > How can i add/modify DPS protocol in SIP message when i creating call? > > > > in pjsua_call_make_call method is parameter msg_data.. > > > > > > pj_status_t pjsua_call_make_call (pjsua_acc_id acc_id, > > const pj_str_t * dst_uri, > > unsigned options, > > void * user_data, > > const pjsua_msg_data * msg_data, > > pjsua_call_id * p_call_id > > ) > > and i suppose this parameter i have to adjust. But how? > > when i try: > > > > pjsua_msg_data msg_data; > > > > pjsua_msg_data_init( &msg_data ); > > > > pj_str_t hname = pj_str("application/sdp"); > > pj_str_t hvalue = pj_str("a=sendonly"); > > > > msg_data.content_type = hname; > > msg_data.msg_body = hvalue; > > > > pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id); > > > > in SIP message is still: > > > > "a=sendrecv" > > > > struct "pjsua_msg_data" have another member "hdr_list" of type > > "pjsip_hdr" Do i have to adjust this member too?..if yes how? > > > > > > thanks > > > > > > > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip@lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org
PL
Peter Lukac
Tue, Apr 27, 2010 2:37 PM

hi

I found point where sdp message is creating.. is it in
endpoint.c source file  and method "pjmedia_endpt_create_sdp"

here is code which add "sendrecv" attribute to SDP message:

/* Add sendrecv attribute. */
attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr);
attr->name = STR_SENDRECV;
m->attr[m->attr_count++] = attr;

as you see attribute "sendrecv" is still added. No condition if add
"sendonly", recvonly" or "sendrecv" attribute should be added :(

Maybe exist some callback which can overwrite? But i can't found anything...

..maybe call pjsua_call_set_hold() will be only solution :)

On Tuesday 27 April 2010 01:32:50 pm Peter Lukac wrote:

hi

thanks for reply.. this trick is little hack :) and i preffer clean
sollutions...it is posible with struct "pjsua_msg_data"? or some callback?
or some another way? ( use pjsua API is not necessary  but welcome :) )

I want control rtp stream all directions (sendonly,recvonly and sendrecv).

thanks

Hi Peter,

On pjsua level, this may be easily done using this trick:

  1. normal make call, pjsua_call_make_call()
  2. once the call gets confirmed, put the call on-hold,
    pjsua_call_set_hold(), call hold will put 'sendonly' direction
    attribute in the sdp.

BR,
nanang

On Mon, Apr 26, 2010 at 7:23 PM, Peter Lukac p.lukac@emtest.sk wrote:

hello again

How can i add/modify DPS protocol in SIP message when i creating call?

in pjsua_call_make_call method is parameter msg_data..

pj_status_t pjsua_call_make_call (pjsua_acc_id acc_id,
const pj_str_t * dst_uri,
unsigned options,
void * user_data,
const pjsua_msg_data * msg_data,
pjsua_call_id * p_call_id
)
and i suppose this parameter i have to adjust. But how?
when i try:

pjsua_msg_data msg_data;

pjsua_msg_data_init( &msg_data );

pj_str_t hname = pj_str("application/sdp");
pj_str_t hvalue = pj_str("a=sendonly");

msg_data.content_type = hname;
msg_data.msg_body = hvalue;

pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id);

in SIP message is still:

"a=sendrecv"

struct "pjsua_msg_data" have another member "hdr_list" of type
"pjsip_hdr" Do i have to adjust this member too?..if yes how?

thanks


Visit our blog: http://blog.pjsip.org

pjsip mailing list
pjsip@lists.pjsip.org
http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org

hi I found point where sdp message is creating.. is it in endpoint.c source file and method "pjmedia_endpt_create_sdp" here is code which add "sendrecv" attribute to SDP message: /* Add sendrecv attribute. */ attr = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_attr); attr->name = STR_SENDRECV; m->attr[m->attr_count++] = attr; as you see attribute "sendrecv" is still added. No condition if add "sendonly", recvonly" or "sendrecv" attribute should be added :( Maybe exist some callback which can overwrite? But i can't found anything... ..maybe call pjsua_call_set_hold() will be only solution :) On Tuesday 27 April 2010 01:32:50 pm Peter Lukac wrote: > hi > > thanks for reply.. this trick is little hack :) and i preffer clean > sollutions...it is posible with struct "pjsua_msg_data"? or some callback? > or some another way? ( use pjsua API is not necessary but welcome :) ) > > I want control rtp stream all directions (sendonly,recvonly and sendrecv). > > thanks > > > Hi Peter, > > > > On pjsua level, this may be easily done using this trick: > > 1. normal make call, pjsua_call_make_call() > > 2. once the call gets confirmed, put the call on-hold, > > pjsua_call_set_hold(), call hold will put 'sendonly' direction > > attribute in the sdp. > > > > BR, > > nanang > > > > On Mon, Apr 26, 2010 at 7:23 PM, Peter Lukac <p.lukac@emtest.sk> wrote: > > > hello again > > > > > > How can i add/modify DPS protocol in SIP message when i creating call? > > > > > > in pjsua_call_make_call method is parameter msg_data.. > > > > > > > > > pj_status_t pjsua_call_make_call (pjsua_acc_id acc_id, > > > const pj_str_t * dst_uri, > > > unsigned options, > > > void * user_data, > > > const pjsua_msg_data * msg_data, > > > pjsua_call_id * p_call_id > > > ) > > > and i suppose this parameter i have to adjust. But how? > > > when i try: > > > > > > pjsua_msg_data msg_data; > > > > > > pjsua_msg_data_init( &msg_data ); > > > > > > pj_str_t hname = pj_str("application/sdp"); > > > pj_str_t hvalue = pj_str("a=sendonly"); > > > > > > msg_data.content_type = hname; > > > msg_data.msg_body = hvalue; > > > > > > pjsua_call_make_call( acc_id, &pj_uri, 0, NULL, &msg_data, &call_id); > > > > > > in SIP message is still: > > > > > > "a=sendrecv" > > > > > > struct "pjsua_msg_data" have another member "hdr_list" of type > > > "pjsip_hdr" Do i have to adjust this member too?..if yes how? > > > > > > > > > thanks > > > > > > > > > > > > > > > _______________________________________________ > > > Visit our blog: http://blog.pjsip.org > > > > > > pjsip mailing list > > > pjsip@lists.pjsip.org > > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > > _______________________________________________ > > Visit our blog: http://blog.pjsip.org > > > > pjsip mailing list > > pjsip@lists.pjsip.org > > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org